Asterisk Reference Information
        Version 1.6.2.14

     Asterisk Development Team
             Asterisk.org
          November 2, 2010



Contents








1 Introduction 8
  1.1 License Information . . . . . . . . . . . . . . . . . . . . . . . . 8
      1.1.1 Hold Music . . . . . . . . . . . . . . . . . . . . . . . . 9

  1.2 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
      1.2.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . 10
      1.2.2 Network Security . . . . . . . . . . . . . . . . . . . . . 10
      1.2.3 Dialplan Security . . . . . . . . . . . . . . . . . . . . . 10

      1.2.4 Log Security . . . . . . . . . . . . . . . . . . . . . . . . 11
  1.3 Hardware . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
      1.3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . 11
      1.3.2 DAHDI compatible hardware . . . . . . . . . . . . . . 12

      1.3.3 Non-DAHDI compatible hardware . . . . . . . . . . . . 13
      1.3.4 mISDN compatible hardware . . . . . . . . . . . . . . 13
      1.3.5 Miscellaneous other interfaces . . . . . . . . . . . . . . 14

2 Configuration 15

  2.1 General Configuration Information . . . . . . . . . . . . . . . 15
      2.1.1 Configuration Parser . . . . . . . . . . . . . . . . . . . 15
      2.1.2 Asterisk.conf . . . . . . . . . . . . . . . . . . . . . . . 19
      2.1.3 CLI Prompt . . . . . . . . . . . . . . . . . . . . . . . . 22

      2.1.4 Extensions . . . . . . . . . . . . . . . . . . . . . . . . . 23
      2.1.5 IP Quality of Service . . . . . . . . . . . . . . . . . . . 24
      2.1.6 MP3 Support . . . . . . . . . . . . . . . . . . . . . . . 28
      2.1.7 ICES . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
  2.2 Database Support . . . . . . . . . . . . . . . . . . . . . . . . . 28


                                    1


       2.2.1 Realtime Database Configuration . . . . . . . . . . . . 28
       2.2.2 FreeTDS . . . . . . . . . . . . . . . . . . . . . . . . . . 31

  2.3 Privacy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
       2.3.1 First of all . . . . . . . . . . . . . . . . . . . . . . . . . 32
       2.3.2 Next, Fight against autodialers!! . . . . . . . . . . . . . 32
       2.3.3 Next, Fight against the empty CALLERID! . . . . . . 32
       2.3.4 Next, use a WELCOME MENU ! . . . . . . . . . . . . 33

       2.3.5 Next: Torture Them! . . . . . . . . . . . . . . . . . . . 34
       2.3.6 Using Call Screening . . . . . . . . . . . . . . . . . . . 35
       2.3.7 The 'N' and 'n' options . . . . . . . . . . . . . . . . . . 36
       2.3.8 Recorded Introductions . . . . . . . . . . . . . . . . . . 37

3 Channel Variables 40
  3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

  3.2 Parameter Quoting . . . . . . . . . . . . . . . . . . . . . . . . 40
  3.3 Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
  3.4 Variable Inheritance . . . . . . . . . . . . . . . . . . . . . . . 41
       3.4.1 Example . . . . . . . . . . . . . . . . . . . . . . . . . . 42
  3.5 Selecting Characters from Variables . . . . . . . . . . . . . . . 42

  3.6 Expressions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
       3.6.1 Spaces Inside Variables Values . . . . . . . . . . . . . . 43
       3.6.2 Operators . . . . . . . . . . . . . . . . . . . . . . . . . 44
       3.6.3 Floating Point Numbers . . . . . . . . . . . . . . . . . 46
       3.6.4 Functions . . . . . . . . . . . . . . . . . . . . . . . . . 47
       3.6.5 Examples . . . . . . . . . . . . . . . . . . . . . . . . . 48

       3.6.6 Numbers Vs. Strings . . . . . . . . . . . . . . . . . . . 50
       3.6.7 Conditionals . . . . . . . . . . . . . . . . . . . . . . . . 50
       3.6.8 Parse Errors . . . . . . . . . . . . . . . . . . . . . . . . 51
       3.6.9 NULL Strings . . . . . . . . . . . . . . . . . . . . . . . 51
       3.6.10 Warning . . . . . . . . . . . . . . . . . . . . . . . . . . 51

       3.6.11 Incompatabilities . . . . . . . . . . . . . . . . . . . . . 52
       3.6.12 Debugging Hints . . . . . . . . . . . . . . . . . . . . . 53
  3.7 Asterisk standard channel variables . . . . . . . . . . . . . . . 55
       3.7.1 Application return values . . . . . . . . . . . . . . . . . 56
       3.7.2 Various application variables . . . . . . . . . . . . . . . 57

       3.7.3 The MeetMe Conference Bridge . . . . . . . . . . . . . 58
       3.7.4 The VoiceMail() application . . . . . . . . . . . . . . . 58
       3.7.5 The VMAuthenticate() application . . . . . . . . . . . 58


                                       2


       3.7.6 DUNDiLookup() . . . . . . . . . . . . . . . . . . . . . 58

       3.7.7 chan dahdi . . . . . . . . . . . . . . . . . . . . . . . . 58
       3.7.8 chan sip . . . . . . . . . . . . . . . . . . . . . . . . . . 59
       3.7.9 chan agent . . . . . . . . . . . . . . . . . . . . . . . . . 59

       3.7.10 The Dial() application . . . . . . . . . . . . . . . . . . 59
       3.7.11 The chanisavail() application . . . . . . . . . . . . . . . 60
       3.7.12 Dialplan Macros . . . . . . . . . . . . . . . . . . . . . . 60

       3.7.13 The ChanSpy() application . . . . . . . . . . . . . . . 60
       3.7.14 OSP . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60

4 AEL: Asterisk Extension Language 62

  4.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
  4.2 Asterisk in a Nutshell . . . . . . . . . . . . . . . . . . . . . . . 63
       4.2.1 Contexts . . . . . . . . . . . . . . . . . . . . . . . . . . 63
       4.2.2 Extensions and priorities . . . . . . . . . . . . . . . . . 63

       4.2.3 Macros . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
       4.2.4 Applications . . . . . . . . . . . . . . . . . . . . . . . . 64
  4.3 Getting Started . . . . . . . . . . . . . . . . . . . . . . . . . . 65

  4.4 Debugging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
  4.5 About "aelparse" . . . . . . . . . . . . . . . . . . . . . . . . . 65
  4.6 General Notes about Syntax . . . . . . . . . . . . . . . . . . . 66

  4.7 Keywords . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
  4.8 Procedural Interface and Internals . . . . . . . . . . . . . . . . 69
       4.8.1 AEL version 2 BNF . . . . . . . . . . . . . . . . . . . . 69

  4.9 AEL Example USAGE . . . . . . . . . . . . . . . . . . . . . . 72
       4.9.1 Comments . . . . . . . . . . . . . . . . . . . . . . . . . 72
       4.9.2 Context . . . . . . . . . . . . . . . . . . . . . . . . . . 73

       4.9.3 Extensions . . . . . . . . . . . . . . . . . . . . . . . . . 73
       4.9.4 Includes . . . . . . . . . . . . . . . . . . . . . . . . . . 74
       4.9.5 #include . . . . . . . . . . . . . . . . . . . . . . . . . . 75
       4.9.6 Dialplan Switches . . . . . . . . . . . . . . . . . . . . . 75

       4.9.7 Ignorepat . . . . . . . . . . . . . . . . . . . . . . . . . 75
       4.9.8 Variables . . . . . . . . . . . . . . . . . . . . . . . . . . 75
       4.9.9 Local Variables . . . . . . . . . . . . . . . . . . . . . . 76

       4.9.10 Loops . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
       4.9.11 Conditionals . . . . . . . . . . . . . . . . . . . . . . . . 77
       4.9.12 Break, Continue, and Return . . . . . . . . . . . . . . 79

       4.9.13 goto, jump, and labels . . . . . . . . . . . . . . . . . . 79


                                        3


       4.9.14 Macros . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

  4.10 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81
  4.11 Semantic Checks . . . . . . . . . . . . . . . . . . . . . . . . . 82
  4.12 Differences with the original version of AEL . . . . . . . . . . 84
  4.13 Hints and Bugs . . . . . . . . . . . . . . . . . . . . . . . . . . 86

  4.14 The Full Power of AEL . . . . . . . . . . . . . . . . . . . . . . 87

5 SLA: Shared Line Appearances 88
  5.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
  5.2 Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . 88

       5.2.1 Summary . . . . . . . . . . . . . . . . . . . . . . . . . 88
       5.2.2 Dialplan . . . . . . . . . . . . . . . . . . . . . . . . . . 88
       5.2.3 Trunks . . . . . . . . . . . . . . . . . . . . . . . . . . . 89

       5.2.4 Stations . . . . . . . . . . . . . . . . . . . . . . . . . . 90
  5.3 Configuration Examples . . . . . . . . . . . . . . . . . . . . . 91
       5.3.1 Basic SLA . . . . . . . . . . . . . . . . . . . . . . . . . 91
       5.3.2 SLA and Voicemail . . . . . . . . . . . . . . . . . . . . 92

  5.4 Call Handling . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
       5.4.1 Summary . . . . . . . . . . . . . . . . . . . . . . . . . 94
       5.4.2 Station goes off hook (not ringing) . . . . . . . . . . . 94

       5.4.3 Station goes off hook (ringing) . . . . . . . . . . . . . . 94
       5.4.4 Line button on a station is pressed . . . . . . . . . . . 95

6 Channel Drivers 96
  6.1 IAX2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

       6.1.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . 96
       6.1.2 Why IAX2? . . . . . . . . . . . . . . . . . . . . . . . . 96
       6.1.3 Configuration . . . . . . . . . . . . . . . . . . . . . . . 98

       6.1.4 IAX2 Jitterbuffer . . . . . . . . . . . . . . . . . . . . . 98
  6.2 mISDN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
       6.2.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . 101
       6.2.2 Features . . . . . . . . . . . . . . . . . . . . . . . . . . 101

       6.2.3 Fast Installation Guide . . . . . . . . . . . . . . . . . . 102
       6.2.4 Pre-Requisites . . . . . . . . . . . . . . . . . . . . . . . 102
       6.2.5 Configuration . . . . . . . . . . . . . . . . . . . . . . . 102
       6.2.6 mISDN CLI commands . . . . . . . . . . . . . . . . . . 105

       6.2.7 mISDN Variables . . . . . . . . . . . . . . . . . . . . . 105
       6.2.8 Debugging and sending bug reports . . . . . . . . . . . 106


                                        4


        6.2.9 Examples . . . . . . . . . . . . . . . . . . . . . . . . . 106
        6.2.10 Known Problems . . . . . . . . . . . . . . . . . . . . . 106
  6.3 Local . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
        6.3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . 106
        6.3.2 Examples . . . . . . . . . . . . . . . . . . . . . . . . . 107

        6.3.3 Trivial Local channel example . . . . . . . . . . . . . . 107
        6.3.4 Delay dialing devices . . . . . . . . . . . . . . . . . . . 109
        6.3.5 Dialing destinations with different information . . . . . 110
        6.3.6 Using callfiles and Local channels . . . . . . . . . . . . 111
        6.3.7 Understanding When To Use /n . . . . . . . . . . . . . 113
        6.3.8 Local channel modifiers . . . . . . . . . . . . . . . . . . 116

7 Distributed Universal Number Discovery (DUNDi) 117
  7.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
  7.2 DUNDIQUERY and DUNDIRESULT . . . . . . . . . . . . . . 118
  7.3 Peering Agreement . . . . . . . . . . . . . . . . . . . . . . . . 118

8 ENUM 132
  8.1 The ENUMLOOKUP dialplan function . . . . . . . . . . . . . 132
        8.1.1 Arguments . . . . . . . . . . . . . . . . . . . . . . . . . 133
        8.1.2 Examples . . . . . . . . . . . . . . . . . . . . . . . . . 134

        8.1.3 Usage notes and subtle features . . . . . . . . . . . . . 136
        8.1.4 Some more Examples . . . . . . . . . . . . . . . . . . . 138

9 AMI: Asterisk Manager Interface 140
  9.1 The Asterisk Manager TCP/IP API . . . . . . . . . . . . . . . 140
  9.2 Command Syntax . . . . . . . . . . . . . . . . . . . . . . . . . 141
  9.3 Manager commands . . . . . . . . . . . . . . . . . . . . . . . . 141
  9.4 Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
  9.5 Ensuring all modules are loaded . . . . . . . . . . . . . . . . . 142
  9.6 Device status reports . . . . . . . . . . . . . . . . . . . . . . . 143
  9.7 Some standard AMI headers . . . . . . . . . . . . . . . . . . . 143
  9.8 Asynchronous Javascript Asterisk Manger (AJAM) . . . . . . 146
        9.8.1 Setup the Asterisk HTTP server . . . . . . . . . . . . . 147
        9.8.2 Allow Manager Access via HTTP . . . . . . . . . . . . 147

        9.8.3 Integration with other web servers . . . . . . . . . . . . 148




                                          5


10 CDR: Call Detail Records 149
   10.1 Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149

   10.2 CDR Fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
   10.3 Variables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
   10.4 Storage Backends . . . . . . . . . . . . . . . . . . . . . . . . . 152

        10.4.1 Microsoft SQL Server . . . . . . . . . . . . . . . . . . . 152
        10.4.2 MySQL . . . . . . . . . . . . . . . . . . . . . . . . . . 155
        10.4.3 PostgreSQL . . . . . . . . . . . . . . . . . . . . . . . . 155
        10.4.4 SQLite 2 . . . . . . . . . . . . . . . . . . . . . . . . . . 156

        10.4.5 SQLite 3 . . . . . . . . . . . . . . . . . . . . . . . . . . 156
        10.4.6 RADIUS . . . . . . . . . . . . . . . . . . . . . . . . . . 156

11 Voicemail 161
   11.1 ODBC Storage . . . . . . . . . . . . . . . . . . . . . . . . . . 161

   11.2 IMAP Storage . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
        11.2.1 Installation Notes . . . . . . . . . . . . . . . . . . . . . 162
        11.2.2 Modify voicemail.conf . . . . . . . . . . . . . . . . . . 164

        11.2.3 IMAP Folders . . . . . . . . . . . . . . . . . . . . . . . 165
        11.2.4 Separate vs. Shared Email Accounts . . . . . . . . . . 165
        11.2.5 IMAP Server Implementations . . . . . . . . . . . . . . 165
        11.2.6 Quota Support . . . . . . . . . . . . . . . . . . . . . . 166

        11.2.7 Application Notes . . . . . . . . . . . . . . . . . . . . . 166

12 SMS 168
   12.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
   12.2 Background . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168

   12.3 Typical use with Asterisk . . . . . . . . . . . . . . . . . . . . 169
   12.4 Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
   12.5 Sub address . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
   12.6 extensions.conf . . . . . . . . . . . . . . . . . . . . . . . . . . 170

   12.7 Using smsq . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
   12.8 File formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
   12.9 Delivery reports . . . . . . . . . . . . . . . . . . . . . . . . . . 179


13 Queues 180
   13.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
   13.2 Configuring Call Queues . . . . . . . . . . . . . . . . . . . . . 180
        13.2.1 queues.conf . . . . . . . . . . . . . . . . . . . . . . . . 180



                                          6


         13.2.2 Routing incoming Calls to Queues . . . . . . . . . . . . 181
         13.2.3 Assigning agents to Queues . . . . . . . . . . . . . . . 184
         13.2.4 Controlling The Way Queues Call the Agents . . . . . 187

         13.2.5 Pre Acknowledgement Message . . . . . . . . . . . . . 189
         13.2.6 Caveats . . . . . . . . . . . . . . . . . . . . . . . . . . 189
   13.3 Queue Logs . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190

14 Phone Provisioning 193
   14.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
   14.2 Configuration of phoneprov.conf . . . . . . . . . . . . . . . . . 193
         14.2.1 The [general] section . . . . . . . . . . . . . . . . . . . 193

         14.2.2 Creating phone profiles . . . . . . . . . . . . . . . . . . 194
   14.3 Configuration of users.conf . . . . . . . . . . . . . . . . . . . . 195
         14.3.1 The [general] section . . . . . . . . . . . . . . . . . . . 195
         14.3.2 Invdividual Users . . . . . . . . . . . . . . . . . . . . . 196
   14.4 Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
   14.5 Putting it all together . . . . . . . . . . . . . . . . . . . . . . 198

15 Packet Loss Concealment 200
   15.1 What is PLC? . . . . . . . . . . . . . . . . . . . . . . . . . . . 200

   15.2 How does Asterisk detect packet loss? . . . . . . . . . . . . . . 200
   15.3 A bit of background on translation . . . . . . . . . . . . . . . 201
   15.4 Additional restrictions and caveats . . . . . . . . . . . . . . . 202
   15.5 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
   15.6 Protip . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203

16 Sounds Packages 205
   16.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 205
   16.2 Getting The Sounds Tools . . . . . . . . . . . . . . . . . . . . 205

   16.3 About The Sounds Tools . . . . . . . . . . . . . . . . . . . . . 206
         16.3.1 audiofilter . . . . . . . . . . . . . . . . . . . . . . . . . 206
         16.3.2 makeg722 . . . . . . . . . . . . . . . . . . . . . . . . . 206
         16.3.3 scripts . . . . . . . . . . . . . . . . . . . . . . . . . . . 207

17 Development 208
   17.1 Backtrace . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208





                                          7




Chapter 1

Introduction


This document contains various pieces of information that are useful for
reference purposes.

1.1 License Information
Asterisk is distributed under the GNU General Public License version 2 and is
also available under alternative licenses negotiated directly with Digium, Inc.
If you obtained Asterisk under the GPL, then the GPL applies to all loadable
Asterisk modules used on your system as well, except as defined below. The
GPL (version 2) is included in this source tree in the file COPYING.
    This package also includes various components that are not part of As-
terisk itself; these components are in the 'contrib' directory and its subdirec-
tories. These components are also distributed under the GPL version 2 as
well.
    Digium, Inc. (formerly Linux Support Services) holds copyright and/or
sufficient licenses to all components of the Asterisk package, and therefore
can grant, at its sole discretion, the ability for companies, individuals, or or-
ganizations to create proprietary or Open Source (even if not GPL) modules
which may be dynamically linked at runtime with the portions of Asterisk
which fall under our copyright/license umbrella, or are distributed under
more flexible licenses than GPL.
    If you wish to use our code in other GPL programs, don't worry -- there
is no requirement that you provide the same exception in your GPL'd prod-


                                             8


ucts (although if you've written a module for Asterisk we would strongly
encourage you to make the same exception that we do).
    Specific permission is also granted to link Asterisk with OpenSSL, OpenH323
and/or the UW IMAP Toolkit and distribute the resulting binary files.
    In addition, Asterisk implements two management/control protocols: the
Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface (AGI).
It is our belief that applications using these protocols to manage or control an
Asterisk instance do not have to be licensed under the GPL or a compatible
license, as we believe these protocols do not create a 'derivative work' as
referred to in the GPL. However, should any court or other judiciary body
find that these protocols do fall under the terms of the GPL, then we hereby
grant you a license to use these protocols in combination with Asterisk in
external applications licensed under any license you wish.
    The 'Asterisk' name and logos are trademarks owned by Digium, Inc., and
use of them is subject to our trademark licensing policies. If you wish to use
these trademarks for purposes other than simple redistribution of Asterisk
source code obtained from Digium, you should contact our licensing depart-
ment to determine the necessary steps you must take. For more information
on this policy, please read:
    http://www.digium.com/en/company/profile/trademarkpolicy.php
    If you have any questions regarding our licensing policy, please contact
us:
    +1.877.344.4861 (via telephone in the USA) +1.256.428.6000 (via tele-
phone outside the USA) +1.256.864.0464 (via FAX inside or outside the
USA) IAX2/pbx.digium.com (via IAX2) licensing@digium.com (via email)
    Digium, Inc. 445 Jan Davis Drive Huntsville, AL 35806 USA

1.1.1 Hold Music
Digium has licensed the music included with the Asterisk distribution From
opsound.org for use and distribution with Asterisk. It is licensed ONLY for
use as hold music within an Asterisk based PBX.





                                             9


1.2 Security

1.2.1 Introduction
PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED
INFORMATION. IMPROPERCONFIGURATIONOFASTERISKCOULD
ALLOWUNAUTHORIZEDUSEOFYOURFACILITIES, POTENTIALLY
INCURRING SUBSTANTIAL CHARGES.
   Asterisk security involves both network security (encryption, authentica-
tion) as well as dialplan security (authorization - who can access services in
your pbx). If you are setting up Asterisk in production use, please make sure
you understand the issues involved.
1.2.2 Network Security

If you install Asterisk and use the "make samples" command to install a
demonstration configuration, Asterisk will open a few ports for accepting
VoIP calls. Check the channel configuration files for the ports and IP ad-
dresses.
   If you enable the manager interface in manager.conf, please make sure
that you access manager in a safe environment or protect it with SSH or
other VPN solutions.
   For all TCP/IP connections in Asterisk, you can set ACL lists that will
permit or deny network access to Asterisk services. Please check the "permit"
and "deny" configuration options in manager.conf and the VoIP channel
configurations - i.e. sip.conf and iax.conf.
   The IAX2 protocol supports strong RSA key authentication as well as
AES encryption of voice and signalling. The SIP channel does not support
encryption in this version of Asterisk.
1.2.3 Dialplan Security
First and foremost remember this:
   USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR
TOLL SERVICES FROM ANY INCOMING CONNECTIONS.
   You should consider that if any channel, incoming line, etc can enter an
extension context that it has the capability of accessing any extension within
that context.

                                            10


    Therefore, you should NOT allow access to outgoing or toll services in
contexts that are accessible (especially without a password) from incoming
channels, be they IAX channels, FX or other trunks, or even untrusted sta-
tions within you network. In particular, never ever put outgoing toll services
in the "default" context. To make things easier, you can include the "default"
context within other private contexts by using:
  include => default
    in the appropriate section. A well designed PBX might look like this:
  [longdistance]
  exten => _91NXXNXXXXXX,1,Dial(DAHDI/g2/${EXTEN:1})
  include => local
  [local]
  exten => _9NXXNXXX,1,Dial(DAHDI/g2/${EXTEN:1})
  include => default
  [default]
  exten => 6123,Dial(DAHDI/1)

    DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR
DEFAULT CONTEXT. There isn't really a security reason, it just will keep
people from wanting to play with your Asterisk setup remotely.
1.2.4 Log Security

Please note that the Asterisk log files, as well as information printed to the
Asterisk CLI, may contain sensitive information such as passwords and call
history. Keep this in mind when providing access to these resources.

1.3 Hardware
1.3.1 Introduction

A PBX is only really useful if you can get calls into it. Of course, you can
use Asterisk with VoIP calls (SIP, H.323, IAX, etc.), but you can also talk
to the real PSTN through various cards.
    Supported Hardware is divided into two general groups: DAHDI de-
vices and non-DAHDI devices. The DAHDI compatible hardware supports
pseudo-TDM conferencing and all call features through chan dahdi, whereas
non-DAHDI compatible hardware may have different features.
                                            11


1.3.2 DAHDI compatible hardware
  * Digium, Inc. (Primary Developer of Asterisk) http://www.digium.com
      -- Analog Interfaces
            * TDM400P - The TDM400P is a half-length PCI 2.2-compliant
              card that supports FXS and FXO station interfaces for con-
              necting analog telephones and analog POTS lines through a
              PC.
            * TDM800P- The TDM800Pis a half-length PCI2.2-compliant,
              8 port card using Digium's VoiceBus technology that supports
              FXS and FXO station interfaces for connecting analog tele-
              phones and analog POTS lines through a PC.
            * TDM2400P- The TDM2400Pis a full-length PCI2.2-compliant
              card for connecting analog telephones and analog POTS lines
              through a PC. It supports a combination of up to 6 FXS
              and/or FXO modules for a total of 24 lines.
      -- Digital Interfaces
            * TE412P - The TE412P offers an on-board DSP-based echo
              cancellation module. It supports E1, T1, and J1 environments
              and is selectable on a per-card or per-port basis.
            * TE410P - The TE410P improves performance and scalability
              through bus mastering architecture. It supports E1, T1, and
              J1 environments and is selectable on a per-card or per-port
              basis.
            * TE407P - The TE407P offers an on-board DSP-based echo
              cancellation module. It supports E1, T1, and J1 environments
              and is selectable on a per-card or per-port basis.
            * TE405P - The TE405P improves performance and scalability
              through bus mastering architecture. It supports both E1, T1,
              J1 environments and is selectable on a per-card or per-port
              basis.
            * TE212P - The TE212P offers an on-board DSP-based echo
              cancellation module. It supports E1, T1, and J1 environments
              and is selectable on a per-card or per-port basis.
            * TE210P - The TE210P improves performance and scalability
              through bus mastering architecture. It supports E1, T1, and

                                       12


               J1 environments and is selectable on a per-card or per-port
               basis.
            * TE207P - The TE207P offers an on-board DSP-based echo
               cancellation module. It supports E1, T1, and J1 environments
               and is selectable on a per-card or per-port basis.
            * TE205P - The TE205P improves performance and scalability
               through bus mastering architecture. It supports both E1 and
               T1/J1 environments and is selectable on a per-card or per-
               port basis.
            * TE120P - The TE120P is a single span, selectable T1, E1,
                                                                 TM
               or J1 card and utilizes Digium's VoiceBus technology. It
               supports both voice and data modes.
            * TE110P- The TE110Pbrings a high-performance, cost-effective,
               and flexible single span togglable T1, E1, J1 interface to the
               Digium line-up of telephony interface devices.

1.3.3 Non-DAHDI compatible hardware
   * QuickNet, Inc. http://www.quicknet.net
       -- Internet PhoneJack - Single FXS interface. Supports Linux tele-
          phony interface. DSP compression built-in.
       -- Internet LineJack - Single FXS or FXO interface. Supports Linux
          telephony interface.

1.3.4 mISDN compatible hardware
mISDN homepage: http://www.misdn.org/
   Any adapter with an mISDN driver should be compatible with chan -
misdn. See the mISDN section for more information.

   * Digium, Inc. (Primary Developer of Asterisk) http://www.digium.com
       -- B410P - 4 Port BRI card (TE/NT)
   * beroNet http://www.beronet.com
       -- BN4S0 - 4 Port BRI card (TE/NT)

                                        13


      -- BN8S0 - 8 Port BRI card (TE/NT)
      -- Billion Card - Single Port BRI card (TE (/NT with crossed cable))

1.3.5 Miscellaneous other interfaces
  * Digium, Inc. (Primary Developer of Asterisk)
      -- TC400B- The TC400Bis a half-length, low-profile PCI2.2-compliant
         card for transforming complex VoIP codecs (G.729) into simple
         codecs.
  * ALSA http://www.alsa-project.org
      -- Any ALSA compatible full-duplex sound card

  * OSS http://www.opensound.com
      -- Any OSS compatible full-duplex sound card















                                     14




Chapter 2

Configuration


2.1 General Configuration Information

2.1.1 Configuration Parser
Introduction
The Asterisk configuration parser in the 1.2 version and beyond series has
been improved in a number of ways. In addition to the realtime architecture,
we now have the ability to create templates in configuration files, and use
these as templates when we configure phones, voicemail accounts and queues.
   These changes are general to the configuration parser, and works in all
configuration files.

General syntax
Asterisk configuration files are defined as follows:
  [section]
  label = value
  label2 = value

   In some files, (e.g. mgcp.conf, dahdi.conf and agents.conf), the syntax is
a bit different. In these files the syntax is as follows:
  [section]
  label1 = value1
  label2 = value2
  object => name
                                           15


   label3 = value3
   label2 = value4
   object2 => name2
    In this syntax, we create objects with the settings defined above the object
creation. Note that settings are inherited from the top, so in the example
above object2 has inherited the setting for "label1" from the first object.
    For template configurations, the syntax for defining a section is changed
to:
   [section](options)
   label = value
    The options field is used to define templates, refer to templates and hide
templates. Any object can be used as a template.
    No whitespace is allowed between the closing "]" and the parenthesis "(".

Comments
All lines that starts with semi-colon ";" is treated as comments and is not
parsed.
    The ";--" is a marker for a multi-line comment. Everything after that
marker will be treated as a comment until the end-marker "--;" is found.
Parsing begins directly after the end-marker.
   ;This is a comment
   label = value
   ;-- This is
   a comment --;
   ;-- Comment --; exten=> 1000,1,dial(SIP/lisa)

Including other files
In all of the configuration files, you may include the content of another file
with the #include statement. The content of the other file will be included
at the row that the #include statement occurred.
   #include myusers.conf

    You may also include the output of a program with the #exec directive,
if you enable it in asterisk.conf
    In asterisk.conf, add the execincludes = yes statement in the options
section:

                                             16


  [options]
  execincludes=yes
   The exec directive is used like this:
  #exec /usr/local/bin/myasteriskconfigurator.sh

Adding to an existing section
  [section]
  label = value
  [section](+)
  label2 = value2
   In this case, the plus sign indicates that the second section (with the
same name) is an addition to the first section. The second section can be in
another file (by using the #include statement). If the section name referred
to before the plus is missing, the configuration will fail to load.

Defining a template-only section
  [section](!)
  label = value

   The exclamation mark indicates to the config parser that this is a only
a template and should not itself be used by the Asterisk module for config-
uration. The section can be inherited by other sections (see section "Using
templates" below) but is not used by itself.
Using templates (or other configuration sections)
  [section](name[,name])
  label = value

   The name within the parenthesis refers to other sections, either templates
or standard sections. The referred sections are included before the configura-
tion engine parses the local settings within the section as though their entire
contents (and anything they were previously based upon) were included in
the new section. For example consider the following:
  [foo]
  disallow=all
  allow=ulaw
  allow=alaw

                                              17


  [bar]
  allow=gsm
  allow=g729
  permit=192.168.2.1
  [baz](foo,bar)
  type=friend
  permit=192.168.3.1
  context=incoming
  host=bnm

   The [baz] section will be processed as though it had been written in the
following way:
  [baz]
  disallow=all
  allow=ulaw
  allow=alaw
  allow=gsm
  allow=g729
  permit=192.168.2.1
  type=friend
  permit=192.168.3.1
  context=incoming
  host=bnm

   It should also be noted that there are no guaranteed overriding semantics,
meaning that if you define something in one template, you should not expect
to be able to override it by defining it again in another template.

Additional Examples

(in top-level sip.conf)
  [defaults](!)
  type=friend
  nat=yes
  qualify=on
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  #include accounts/*/sip.conf

   (in accounts/customer1/sip.conf)
  [def-customer1](!,defaults)
  secret=this_is_not_secret
  context=from-customer1
  callerid=Customer 1 <300>
  accountcode=0001

                                          18


  [phone1](def-customer1)
  mailbox=phone1@customer1
  [phone2](def-customer1)
  mailbox=phone2@customer1

   This example defines two phones - phone1 and phone2 with settings inher-
ited from "def-customer1". The "def-customer1" is a template that inherits
from "defaults", which also is a template.

2.1.2 Asterisk.conf

Asterisk Main Configuration File

Below is a sample of the main Asterisk configuration file, asterisk.conf. Note
that this file is not provided in sample form, because the Makefile creates it
when needed and does not touch it when it already exists.
  [directories]
  ; Make sure these directories have the right permissions if not
  ; running Asterisk as root
  ; Where the configuration files (except for this one) are located
  astetcdir => /etc/asterisk

  ; Where the Asterisk loadable modules are located
  astmoddir => /usr/lib/asterisk/modules
  ; Where additional 'library' elements (scripts, etc.) are located
  astvarlibdir => /var/lib/asterisk
  ; Where AGI scripts/programs are located
  astagidir => /var/lib/asterisk/agi-bin

  ; Where spool directories are located
  ; Voicemail, monitor, dictation and other apps will create files here
  ; and outgoing call files (used with pbx_spool) must be placed here
  astspooldir => /var/spool/asterisk
  ; Where the Asterisk process ID (pid) file should be created
  astrundir => /var/run/asterisk
  ; Where the Asterisk log files should be created
  astlogdir => /var/log/asterisk

  [options]
  ;Under "options" you can enter configuration options
  ;that you also can set with command line options

  ; Verbosity level for logging (-v)

                                             19


verbose = 0
; Debug: "No" or value (1-4)
debug = 3

; Background execution disabled (-f)
nofork=yes | no

; Always background, even with -v or -d (-F)
alwaysfork=yes | no
; Console mode (-c)
console= yes | no

; Execute with high priority (-p)
highpriority = yes | no
; Initialize crypto at startup (-i)
initcrypto = yes | no

; Disable ANSI colors (-n)
nocolor = yes | no
; Dump core on failure (-g)
dumpcore = yes | no

; Run quietly (-q)
quiet = yes | no
; Force timestamping in CLI verbose output (-T)
timestamp = yes | no

; User to run asterisk as (-U) NOTE: will require changes to
; directory and device permissions
runuser = asterisk
; Group to run asterisk as (-G)
rungroup = asterisk

; Enable internal timing support (-I)
internal_timing = yes | no

; Language Options
documentation_language = en | es | ru
; These options have no command line equivalent

; Cache record() files in another directory until completion
cache_record_files = yes | no
record_cache_dir = <dir>
; Build transcode paths via SLINEAR
transcode_via_sln = yes | no

; send SLINEAR silence while channel is being recorded
transmit_silence_during_record = yes | no


                                         20


; The maximum load average we accept calls for
maxload = 1.0
; The maximum number of concurrent calls you want to allow
maxcalls = 255

; Stop accepting calls when free memory falls below this amount specified in MB
minmemfree = 256

; Allow #exec entries in configuration files
execincludes = yes | no
; Don't over-inform the Asterisk sysadm, he's a guru
dontwarn = yes | no

; System name. Used to prefix CDR uniqueid and to fill \${SYSTEMNAME}
systemname = <a_string>
; Should language code be last component of sound file name or first?
; when off, sound files are searched as <path>/<lang>/<file>
; when on, sound files are search as <lang>/<path>/<file>
; (only affects relative paths for sound files)
languageprefix = yes | no

; Locking mode for voicemail
; - lockfile: default, for normal use
; - flock: for where the lockfile locking method doesn't work
; eh. on SMB/CIFS mounts
lockmode = lockfile | flock
; Entity ID. This is in the form of a MAC address. It should be universally
; unique. It must be unique between servers communicating with a protocol
; that uses this value. The only thing that uses this currently is DUNDi,
; but other things will use it in the future.
; entityid=00:11:22:33:44:55

[files]
; Changing the following lines may compromise your security
; Asterisk.ctl is the pipe that is used to connect the remote CLI
; (asterisk -r) to Asterisk. Changing these settings change the
; permissions and ownership of this file.
; The file is created when Asterisk starts, in the "astrundir" above.

;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = asterisk
;astctl = asterisk.ctl









                                         21


2.1.3 CLI Prompt
Changing the CLI Prompt
The CLI prompt is set with the ASTERISK PROMPT UNIX environment
variable that you set from the Unix shell before starting Asterisk
   You may include the following variables, that will be replaced by the
current value by Asterisk:

   * %d - Date (year-month-date)
   * %s - Asterisk system name (from asterisk.conf)
   * %h - Full hostname
   * %H - Short hostname

   * %t - Time
   * %u - Username
   * %g - Groupname
   * %% - Percent sign

   * %# - '#' if Asterisk is run in console mode, '>' if running as remote
      console
   * %Cn[;n] - Change terminal foreground (and optional background) color
      to specified A full list of colors may be found in include/asterisk/
      term.h
   On systems which implement getloadavg(3), you may also use:

   * %l1 - Load average over past minute
   * %l2 - Load average over past 5 minutes
   * %l3 - Load average over past 15 minutes



                                           22


2.1.4 Extensions
The Asterisk dialplan
The Asterisk dialplan is divided into contexts. A context is simply a group
of extensions. For each "line" that should be able to be called, an extension
must be added to a context. Then, you configure the calling "line" to have
access to this context.
    If you change the dialplan, you can use the Asterisk CLI command "di-
alplan reload" to load the new dialplan without disrupting service in your
PBX.
    Extensions are routed according to priority and may be based on any
set of characters (a-z), digits, #, and *. Please note that when matching a
pattern, "N", "X", and "Z" are interpreted as classes of digits.
    For each extension, several actions may be listed and must be given a
unique priority. When each action completes, the call continues at the next
priority (except for some modules which use explicitly GOTO's).
    Extensions frequently have data they pass to the executing application
(most frequently a string). You can see the available dialplan applications
by entering the "core show applications" command in the CLI.
    In this version of Asterisk, dialplan functions are added. These can be
used as arguments to any application. For a list of the installed functions in
your Asterisk, use the "core show functions" command.

Example dialplan
The example dial plan, in the configs/extensions.conf.sample file is in-
stalled as extensions.conf if you run "make samples" after installation of
Asterisk. This file includes many more instructions and examples than this
file, so it's worthwhile to read it.

Special extensions
There are some extensions with important meanings:

    * s
          -- What to do when an extension context is entered (unless over-
              ridden by the low level channel interface) This is used in macros,

                                            23


            and some special cases. "s" is not a generic catch-all wildcard
            extension.
   * i
         -- What to do if an invalid extension is entered

   * h
         -- The hangup extension, executed at hangup
   * t

         -- What to do if nothing is entered in the requisite amount of time.
   * T
         -- This is the extension that is executed when the 'absolute' time-
            out is reached. See "core show function TIMEOUT" for more
            information on setting timeouts.
   * e

         -- This extension will substitute as a catchall for any of the 'i', 't',
            or 'T' extensions, if any of them do not exist and catching the
            error in a single routine is desired. The function EXCEPTION
            may be used to query the type of exception or the location where
            it occurred.
   And finally, the extension context "default" is used when either a) an
extension context is deleted while an extension is in use, or b) a specific
starting extension handler has not been defined (unless overridden by the
low level channel interface).

2.1.5 IP Quality of Service
Introduction
Asterisk supports different QoS settings at the application level for various
protocols on both signaling and media. The Type of Service (TOS) byte can
be set on outgoing IP packets for various protocols. The TOS byte is used

                                           24


by the network to provide some level of Quality of Service (QoS) even if the
network is congested with other traffic.
    Asterisk running on Linux can also set 802.1p CoS marks in VLAN pack-
ets for the VoIP protocols it uses. This is useful when working in a switched
environment. In fact Asterisk only set priority for Linux socket. For mapping
this priority and VLAN CoS mark you need to use this command:
vconfig set_egress_map [vlan-device] [skb-priority] [vlan-qos]

    The table below shows all VoIP channel drivers and other Asterisk mod-
ules that support QoS settings for network traffic. It also shows the type(s)
of traffic for which each module can support setting QoS settings.
                                  Signaling Audio Video Text
                 chan sip + + + +
                 chan skinny + + +
                 chan mgcp + +
                 chan unistm + +
                 chan h323 +
                 chan iax2 +

                     Table 2.1: Channel Driver QoS Settings

                                  Signaling Audio Video Text
                 dundi.conf + (tos setting)
                 iaxprov.conf + (tos setting)

                          Table 2.2: Other ToS Settings


IP TOS values
The allowable values for any of the tos* parameters are: CS0, CS1, CS2,
CS3, CS4, CS5, CS6, CS7, AF11, AF12, AF13, AF21, AF22, AF23, AF31,
AF32, AF33, AF41, AF42, AF43 and ef (expedited forwarding),
    The tos* parameters also take numeric values.
                                          25


   Note that on a Linux system, Asterisk must be compiled with libcap in
order to use the ef tos setting if Asterisk is not run as root.
   The lowdelay, throughput, reliability, mincost, and none values have been
removed in current releases.

802.1p CoS values
Because 802.1p uses 3 bits of the VLAN header, this parameter can take
integer values from 0 to 7.
Recommended values

The recommended values shown below are also included in sample configu-
ration files:
                                             tos cos
                              Signaling cs3 3
                              Audio ef 5
                              Video af41 4
                              Text af41 3
                              Other ef

                    Table 2.3: Recommended QoS Settings

IAX2

In iax.conf, there is a "tos" parameter that sets the global default TOS
for IAX packets generated by chan iax2. Since IAX connections combine
signalling, audio, and video into one UDP stream, it is not possible to set
the TOS separately for the different types of traffic.
   In iaxprov.conf, there is a "tos" parameter that tells the IAXy what
TOS to set on packets it generates. As with the parameter in iax.conf, IAX
packets generated by an IAXy cannot have different TOS settings based upon
the type of packet. However different IAXy devices can have different TOS
settings.


                                          26


SIP
In sip.conf, there are four parameters that control the TOS settings: "tos -
sip", "tos audio", "tos video" and "tos text". tos sip controls what TOS SIP
call signaling packets are set to. tos audio, tos video and tos text control
what TOS values are used for RTP audio, video, and text packets, respec-
tively.
    There are four parameters to control 802.1p CoS: "cos sip", "cos audio",
"cos video" and "cos text". The behavior of these parameters is the same as
for the SIP TOS settings described above.

Other RTP channels
chan mgcp, chan h323, chan skinny and chan unistim also support TOS and
CoS via setting tos and cos parameters in their corresponding configuration
files. Naming style and behavior are the same as for chan sip.

Reference
IEEE802.1QStandard: http://standards.ieee.org/getieee802/download/802.1Q-1998.pdf
Related protocols: IEEE 802.3, 802.2, 802.1D, 802.1Q
    RFC 2474 - "Definition of the Differentiated Services Field (DS field) in
the IPv4 and IPv6 Headers", Nichols, K., et al, December 1998.
    IANA Assignments, DSCP registry Differentiated Services Field Code-
points http://www.iana.org/assignments/dscp-registry
    To get the most out of setting the TOS on packets generated by Asterisk,
you will need to ensure that your network handles packets with a TOS prop-
erly. For Cisco devices, see the previously mentioned "Enterprise QoS Solu-
tion Reference Network Design Guide". For Linux systems see the "Linux
Advanced Routing & Traffic Control HOWTO" at http://www.lartc.org/.
    For more information on Quality of Service for VoIP networks see the
"Enterprise QoS Solution Reference Network Design Guide" version 3.3 from
Cisco at: http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmigration_09186a008049b062.pdf





                                         27


2.1.6 MP3 Support
MP3 Music On Hold
Use of the mpg123 for your music on hold is no longer recommended and is
now officially deprecated. You should now use one of the native formats for
your music on hold selections.
    However, if you still need to use mp3 as your music on hold format,
a format driver for reading MP3 audio files is available in the asterisk-
addons SVN repository on svn.digium.com or in the asterisk-addons release
at http://downloads.digium.com/pub/telephony/asterisk/.

2.1.7 ICES
The advent of icecast into Asterisk allows you to do neat things like have a
caller stream right into an ice-cast stream as well as using chan local to place
things like conferences, music on hold, etc. into the stream.
    You'll need to specify a config file for the ices encoder. An example is
included in contrib/asterisk-ices.xml.

2.2 Database Support

2.2.1 Realtime Database Configuration
Introduction
The Asterisk Realtime Architecture is a new set of drivers and functions
implemented in Asterisk.
    The benefits of this architecture are many, both from a code management
standpoint and from an installation perspective.
    The ARA is designed to be independent of storage. Currently, most
drivers are based on SQL, but the architecture should be able to handle
other storage methods in the future, like LDAP.
    The main benefit comes in the database support. In Asterisk v1.0 some
functions supported MySQL database, some PostgreSQL and other ODBC.
With the ARA, we have a unified database interface internally in Asterisk,
so if one function supports database integration, all databases that has a
realtime driver will be supported in that function.
    Currently there are three realtime database drivers:

                                           28


    * ODBC: Support for UnixODBC, integrated into Asterisk The UnixODBC
        subsystem supports many different databases, please check www.unixodbc.org
        for more information.
    * MySQL: Found in the asterisk-addons subversion repository on svn.digium.com

    * PostgreSQL: Native support for Postgres, integrated into Asterisk
Two modes: Static and Realtime
The ARA realtime mode is used to dynamically load and update objects.
This mode is used in the SIP and IAX2 channels, as well as in the voicemail
system. For SIP and IAX2 this is similar to the v1.0 MYSQL FRIENDS
functionality. With the ARA, we now support many more databases for
dynamic configuration of phones.
    The ARA static mode is used to load configuration files. For the Asterisk
modules that read configurations, there's no difference between a static file
in the file system, like extensions.conf, and a configuration loaded from a
database.
    You just have to always make sure the var metric values are properly set
and ordered as you expect in your database server if you're using the static
mode with ARA (either sequentially or with the same var metric value for
everybody).
    If you have an option that depends on another one in a given configuration
file (i.e, 'musiconhold' depending on 'agent' from agents.conf) but their var -
metric are not sequential you'll probably get default values being assigned
for those options instead of the desired ones. You can still use the same
var metric for all entries in your DB, just make sure the entries are recorded
in an order that does not break the option dependency.
    That doesn't happen when you use a static file in the file system. Al-
though this might be interpreted as a bug or limitation, it is not.

Realtime SIP friends
The SIP realtime objects are users and peers that are loaded in memory
when needed, then deleted. This means that Asterisk currently can't handle
voicemail notification and NAT keepalives for these peers. Other than that,
most of the functionality works the same way for realtime friends as for the
ones in static configuration.

                                             29


   With caching, the device stays in memory for a specified time. More
information about this is to be found in the sip.conf sample file.
   If you specify a separate family called "sipregs" SIP registration data will
be stored in that table and not in the "sippeers" table.

Realtime H.323 friends
Like SIP realtime friends, H.323 friends also can be configured using dynamic
realtime objects.
New function in the dial plan: The Realtime Switch

The realtime switch is more than a port of functionality in v1.0 to the new
architecture, this is a new feature of Asterisk based on the ARA. The real-
time switch lets your Asterisk server do database lookups of extensions in
realtime from your dial plan. You can have many Asterisk servers sharing a
dynamically updated dial plan in real time with this solution.
   Note that this switch does NOT support Caller ID matching, only exten-
sion name or pattern matching.
Capabilities
The realtime Architecture lets you store all of your configuration in databases
and reload it whenever you want. You can force a reload over the AMI,
Asterisk Manager Interface or by calling Asterisk from a shell script with
   asterisk -rx "reload"
   You may also dynamically add SIP and IAX devices and extensions and
making them available without a reload, by using the realtime objects and
the realtime switch.

Configuration in extconfig.conf
You configure the ARA in extconfig.conf (yes, it's a strange name, but is was
defined in the early days of the realtime architecture and kind of stuck).
   The part of Asterisk that connects to the ARA use a well defined family
name to find the proper database driver. The syntax is easy:

     <family> => <realtime driver>,<db name>[,<table>]

                                           30


   The options following the realtime driver identified depends on the driver.
   Defined well-known family names are:
   * sippeers, sipusers - SIP peers and users

   * iaxpeers, iaxusers - IAX2 peers and users
   * voicemail - Voicemail accounts
   * queues - Queues

   * queue members - Queue members
   * extensions - Realtime extensions (switch)
   Voicemail storage with the support of ODBC described in file docs/
odbcstorage.tex (11.1).

Limitations
Currently, realtime extensions do not support realtime hints. There is a
workaround available by using func odbc. See the sample func odbc.conf for
more information.

FreeTDS supported with connection pooling
In order to use a FreeTDS-based database with realtime, you need to turn
connection pooling on in res odbc.conf. This is due to a limitation within
the FreeTDS protocol itself. Please note that this includes databases such as
MS SQL Server and Sybase. This support is new in the current release.

2.2.2 FreeTDS
The cdr tds module now works with most modern release versions of FreeTDS
(from at least 0.60 through 0.82). Although versions of FreeTDS prior to 0.82
will work, we recommend using the latest available version for performance
and stability reasons.
   The latest release of FreeTDS is available from http://www.freetds.org/



                                           31


2.3 Privacy
So, you want to avoid talking to pesky telemarketers/charity seekers/poll
takers/magazine renewers/etc?

2.3.1 First of all
the FTC "Don't call" database, this alone will reduce your telemarketing call
volume considerably. (see: https://www.donotcall.gov/default.aspx ) But,
this list won't protect from the Charities, previous business relationships, etc.

2.3.2 Next, Fight against autodialers!!
Zapateller detects if callerid is present, and if not, plays the da-da-da tones
that immediately precede messages like, "I'm sorry, the number you have
called is no longer in service."
    Most humans, even those with unlisted/callerid-blocked numbers, will not
immediately slam the handset down on the hook the moment they hear the
three tones. But autodialers seem pretty quick to do this.
    I just counted 40 hangups in Zapateller over the last year in my CDR's.
So, that is possibly 40 different telemarketers/charities that have hopefully
slashed my back-waters, out-of-the-way, humble home phone number from
their lists.
    I highly advise Zapateller for those seeking the nirvana of "privacy".

2.3.3 Next, Fight against the empty CALLERID!
A considerable percentage of the calls you don't want, come from sites that
do not provide CallerID.
    Null callerid's are a fact of life, and could be a friend with an unlisted
number, or some charity looking for a handout. The PrivacyManager appli-
cation can help here. It will ask the caller to enter a 10-digit phone number.
They get 3 tries(configurable), and this is configurable, with control being
passed to next priority where you can check the channelvariable PRIVA-
CYMGRSTATUS. If the callerid was valid this variable will have the value
SUCCESS, otherwise it will have the value FAILED.
    PrivacyManager can't guarantee that the number they supply is any good,
tho, as there is no way to find out, short of hanging up and calling them back.

                                              32


But some answers are obviously wrong. For instance, it seems a common
practice for telemarketers to use your own number instead of giving you
theirs. A simple test can detect this. More advanced tests would be to look
for -555- numbers, numbers that count up or down, numbers of all the same
digit, etc.
    PrivacyManager can be told about a context where you can have patterns
that describe valid phone numbers. If none of the patterns match the input,
it will be considered a non-valid phonenumber and the user can try again
until the retry counter is reached. This helps in resolving the issues stated
in the previous paragraph.
    My logs show that 39 have hung up in the PrivacyManager script over
the last year.
    (Note: Demanding all unlisted incoming callers to enter their CID may
not always be appropriate for all users. Another option might be to use call
screening. See below.)

2.3.4 Next, use a WELCOME MENU !
Experience has shown that simply presenting incoming callers with a set of
options, no matter how simple, will deter them from calling you. In the vast
majority of situations, a telemarketer will simply hang up rather than make
a choice and press a key.
    This will also immediately foil all autodialers that simply belch a message
in your ear and hang up.

Example usage of Zapateller and PrivacyManager
  [homeline]
  exten => s,1,Answer
  exten => s,2,SetVar,repeatcount=0
  exten => s,3,Zapateller,nocallerid
  exten => s,4,PrivacyManager
       ;; do this if they don't enter a number to Privacy Manager
  exten => s,5,GotoIf($[ "${PRIVACYMGRSTATUS}" = "FAILED" ]?s,105)
  exten => s,6,GotoIf($[ "${CALLERID(num)}" = "7773334444" & "${CALLERID(name)}" : "Privacy Manager" ]?callerid-liar,s,1:s,7)
  exten => s,7,Dial(SIP/yourphone)
  exten => s,105,Background(tt-allbusy)
  exten => s,106,Background(tt-somethingwrong)
  exten => s,107,Background(tt-monkeysintro)
  exten => s,108,Background(tt-monkeys)
  exten => s,109,Background(tt-weasels)
  exten => s,110,Hangup


                                             33


    I suggest using Zapateller at the beginning of the context, before anything
else, on incoming calls.This can be followed by the PrivacyManager App.
    Make sure, if you do the PrivacyManager app, that you take care of the
error condition! or their non-compliance will be rewarded with access to the
system. In the above, if they can't enter a 10-digit number in 3 tries, they get
the humorous "I'm sorry, but all household members are currently helping
other telemarketers...", "something is terribly wrong", "monkeys have carried
them away...", various loud monkey screechings, "weasels have...", and a
hangup. There are plenty of other paths to my torture scripts, I wanted to
have some fun.
    In nearly all cases now, the telemarketers/charity-seekers that usually get
thru to my main intro, hang up. I guess they can see it's pointless, or the
average telemarketer/charity-seeker is instructed not to enter options when
encountering such systems. Don't know.

2.3.5 Next: Torture Them!
I have developed an elaborate script to torture Telemarketers, and entertain
friends. (See http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture
)
    While mostly those that call in and traverse my teletorture scripts are
those we know, and are doing so out of curiosity, there have been these
others from Jan 1st,2004 thru June 1st, 2004: (the numbers may or may not
be correct.)
    * 603890zzzz -- hung up telemarket options.

    * "Integrated Sale" -- called a couple times. hung up in telemarket options
    * "UNITED STATES GOV" -- maybe a military recruiter, trying to lure
       one of my sons.
    * 800349zzzz -- hung up in charity intro

    * 800349zzzz -- hung up in charity choices, intro, about the only one who
       actually travelled to the bitter bottom of the scripts!
    * 216377zzzz -- hung up the magazine section


                                               34


   * 626757zzzz = "LIR " (pronounced "Liar"?) hung up in telemarket
      intro, then choices
   * 757821zzzz -- hung up in new magazine subscription options.

   That averages out to maybe 1 a month. That puts into question whether
the ratio of the amount of labor it took to make the scripts versus the benefits
of lower call volumes was worth it, but, well, I had fun, so what the heck.
   but, that's about it. Not a whole lot. But I haven't had to say "NO" or
"GO AWAY" to any of these folks for about a year now ...!

2.3.6 Using Call Screening
Another option is to use call screening in the Dial command. It has two main
privacy modes, one that remembers the CID of the caller, and how the callee
wants the call handled, and the other, which does not have a "memory".
   Turning on these modes in the dial command results in this sequence of
events, when someone calls you at an extension:
   1. The caller calls the Asterisk system, and at some point, selects an
      option or enters an extension number that would dial your extension.

   2. Before ringing your extension, the caller is asked to supply an intro-
      duction. The application asks them: "After the tone, say your name".
      They are allowed 4 seconds of introduction.
   3. After that, they are told "Hang on, we will attempt to connect you
      to your party. Depending on your dial options, they will hear ringing
      indications, or get music on hold. I suggest music on hold.
   4. Your extension is then dialed. When (and if) you pick up, you are told
      that a caller presenting themselves as <their recorded intro is played>
      is calling, and you have options, like being connected, sending them to
      voicemail, torture, etc.

   5. You make your selection, and the call is handled as you chose.
   There are some variations, and these will be explained in due course.
   To use these options, set your Dial to something like:

                                             35


  exten => 3,3,Dial(DAHDI/5r3&DAHDI/6r3,35,tmPA(beep))
      or
  exten => 3,3,Dial(DAHDI/5r3&DAHDI/6r3,35,tmP(something)A(beep))
      or
  exten => 3,3,Dial(DAHDI/5r3&DAHDI/6r3,35,tmpA(beep))
    The 't' allows the dialed party to transfer the call using '#'. It's optional.
    The 'm' is for music on hold. I suggest it. Otherwise, the calling party
gets to hear all the ringing, and lack thereof. It is generally better to use
Music On Hold. Lots of folks hang up after the 3rd or 4th ring, and you
might lose the call before you can enter an option!
    The 'P' option alone will database everything using the extension as a
default 'tree'. To get multiple extensions sharing the same database, use
P(some-shared-key). Also, if the same person has multiple extensions, use
P(unique-id) on all their dial commands.
    Use little 'p' for screening. Every incoming call will include a prompt for
the callee's choice.
    the A(beep), will generate a 'beep' that the callee will hear if they choose
to talk to the caller. It's kind of a prompt to let the callee know that he has
to say 'hi'. It's not required, but I find it helpful.
    When there is no CallerID, P and p options will always record an in-
tro for the incoming caller. This intro will be stored temporarily in the
/var/lib/asterisk/sounds/priv-callerintros dir, under the name NO-
CALLERID <extension> <channelname> and will be erased after the callee
decides what to do with the call.
    Of course, NOCALLERID is not stored in the database. All those with
no CALLERID will be considered "Unknown".

2.3.7 The 'N' and 'n' options
Two other options exist, that act as modifiers to the privacy options 'P' and
'p'. They are 'N' and 'n'. You can enter them as dialing options, but they
only affect things if P or p are also in the options.
    'N' says, "Only screen the call if no CallerID is present". So, if a callerID
were supplied, it will come straight thru to your extension.
    'n' says, "Don't save any introductions". Folks will be asked to supply
an introduction ("At the tone, say your name") every time they call. Their
introductions will be removed after the callee makes a choice on how to handle
the call. Whether the P option or the p option is used, the incoming caller
will have to supply their intro every time they call.

                                              36


2.3.8 Recorded Introductions
Philosophical Side Note

The 'P' option stores the CALLERID in the database, along with the callee's
choice of actions, as a convenience to the CALLEE, whereas introductions
are stored and re-used for the convenience of the CALLER.
Introductions
Unless instructed to not save introductions (see the 'n' option above), the
screening modes will save the recordings of the caller's names in the directory
/var/lib/asterisk/sounds/priv-callerintros, if they have a CallerID.
Just the 10-digit callerid numbers are used as filenames, with a ".gsm" at
the end.
    Having these recordings around can be very useful, however...
    First of all, if a callerid is supplied, and a recorded intro for that number
is already present, the caller is spared the inconvenience of having to supply
their name, which shortens their call a bit.
    Next of all, these intros can be used in voicemail, played over loudspeak-
ers, and perhaps other nifty things. For instance:

   exten => s,6,Set(PATH=/var/lib/asterisk/sounds/priv-callerintros)
   exten => s,7,System(/usr/bin/play ${PATH}/${CALLERID(num)}.gsm&,0)
    When a call comes in at the house, the above priority gets executed, and
the callers intro is played over the phone systems speakers. This gives us a
hint who is calling.
    (Note: the ,0 option at the end of the System command above, is a local
mod I made to the System command. It forces a 0 result code to be returned,
whether the play command successfully completed or not. Therefore, I don't
have to ensure that the file exists or not. While I've turned this mod into
the developers, it hasn't been incorporated yet. You might want to write an
AGI or shell script to handle it a little more intelligently)
    And one other thing. You can easily supply your callers with an option
to listen to, and re-record their introductions. Here's what I did in the home
system's extensions.conf. (assume that a Goto(home-introduction,s,1) exists
somewhere in your main menu as an option):
   [home-introduction]
   exten => s,1,Background(intro-options) ;; Script:

                                              37


     ;; To hear your Introduction, dial 1.
     ;; to record a new introduction, dial 2.
     ;; to return to the main menu, dial 3.
     ;; to hear what this is all about, dial 4.
exten => 1,1,Playback,priv-callerintros/${CALLERID(num)}
exten => 1,2,Goto(s,1)
exten => 2,1,Goto(home-introduction-record,s,1)
exten => 3,1,Goto(homeline,s,7)
exten => 4,1,Playback(intro-intro)
     ;; Script:
     ;; This may seem a little strange, but it really is a neat
     ;; thing, both for you and for us. I've taped a short introduction
     ;; for many of the folks who normally call us. Using the Caller ID
     ;; from each incoming call, the system plays the introduction
     ;; for that phone number over a speaker, just as the call comes in.
     ;; This helps the folks
     ;; here in the house more quickly determine who is calling.
     ;; and gets the right ones to gravitate to the phone.
     ;; You can listen to, and record a new intro for your phone number
     ;; using this menu.
exten => 4,2,Goto(s,1)
exten => t,1,Goto(s,1)
exten => i,1,Background(invalid)
exten => i,2,Goto(s,1)
exten => o,1,Goto(s,1)
[home-introduction-record]
exten => s,1,Background(intro-record-choices) ;; Script:
     ;; If you want some advice about recording your
     ;; introduction, dial 1.
     ;; otherwise, dial 2, and introduce yourself after
     ;; the beep.
exten => 1,1,Playback(intro-record)
     ;; Your introduction should be short and sweet and crisp.
     ;; Your introduction will be limited to 4 seconds.
     ;; This is NOT meant to be a voice mail message, so
     ;; please, don't say anything about why you are calling.
     ;; After we are done making the recording, your introduction
     ;; will be saved for playback.
     ;; If you are the only person that would call from this number,
     ;; please state your name. Otherwise, state your business
     ;; or residence name instead. For instance, if you are
     ;; friend of the family, say, Olie McPherson, and both
     ;; you and your kids might call here a lot, you might
     ;; say: "This is the distinguished Olie McPherson Residence!"
     ;; If you are the only person calling, you might say this:
     ;; "This is the illustrious Kermit McFrog! Pick up the Phone, someone!!"
     ;; If you are calling from a business, you might pronounce a more sedate introduction,like,
     ;; "Fritz from McDonalds calling.", or perhaps the more original introduction:
     ;; "John, from the Park County Morgue. You stab 'em, we slab 'em!".
     ;; Just one caution: the kids will hear what you record every time
     ;; you call. So watch your language!
     ;; I will begin recording after the tone.
     ;; When you are done, hit the # key. Gather your thoughts and get
     ;; ready. Remember, the # key will end the recording, and play back
     ;; your intro. Good Luck, and Thank you!"
exten => 1,2,Goto(2,1)


                                         38


  exten => 2,1,Background(intro-start)
       ;; OK, here we go! After the beep, please give your introduction.
  exten => 2,2,Background(beep)
  exten => 2,3,Record(priv-callerintros/${CALLERID(num)}:gsm,4)
  exten => 2,4,Background(priv-callerintros/${CALLERID(num)})
  exten => 2,5,Goto(home-introduction,s,1)
  exten => t,1,Goto(s,1)
  exten => i,1,Background(invalid)
  exten => i,2,Goto(s,1)
  exten => o,1,Goto(s,1)

   In the above, you'd most likely reword the messages to your liking, and
maybe do more advanced things with the 'error' conditions (i,o,t priorities),
but I hope it conveys the idea.




























                                            39




Chapter 3

Channel Variables


3.1 Introduction

There are two levels of parameter evaluation done in the Asterisk dial plan
in extensions.conf.
   1. The first, and most frequently used, is the substitution of variable ref-
      erences with their values.
   2. Then there are the evaluations of expressions done in $[ .. ]. This will
      be discussed below.
Asterisk has user-defined variables and standard variables set by various
modules in Asterisk. These standard variables are listed at the end of this
document.

3.2 Parameter Quoting

  exten => s,5,BackGround,blabla
The parameter (blabla) can be quoted ("blabla"). In this case, a comma does
not terminate the field. However, the double quotes will be passed down to
the Background command, in this example.
   Also, characters special to variable substitution, expression evaluation,
etc (see below), can be quoted. For example, to literally use a $ on the string

                                          40


"$1231", quote it with a preceding \. Special characters that must be quoted
to be used, are [ ] $ " \. (to write \itself, use \).
   These Double quotes and escapes are evaluated at the level of the asterisk
config file parser.
   Double quotes can also be used inside expressions, as discussed below.

3.3 Variables

Parameter strings can include variables. Variable names are arbitrary strings.
They are stored in the respective channel structure.
   To set a variable to a particular value, do:
      exten => 1,2,Set(varname=value)
You can substitute the value of a variable everywhere using ${variablename}.
For example, to stringwise append $lala to $blabla and store result in $koko,
do:
     exten => 1,2,Set(koko=${blabla}${lala})
   There are two reference modes - reference by value and reference by name.
To refer to a variable with its name (as an argument to a function that
requires a variable), just write the name. To refer to the variable's value,
enclose it inside ${}. For example, Set takes as the first argument (before
the =) a variable name, so:
    exten => 1,2,Set(koko=lala)
    exten => 1,3,Set(${koko}=blabla)

stores to the variable "koko" the value "lala" and to variable "lala" the value
"blabla".
   In fact, everything contained ${here} is just replaced with the value of
the variable "here".

3.4 Variable Inheritance
Variable names which are prefixed by " " will be inherited to channels that
are created in the process of servicing the original channel in which the
variable was set. When the inheritance takes place, the prefix will be removed
in the channel inheriting the variable. If the name is prefixed by " " in the

                                            41


channel, then the variable is inherited and the " " will remain intact in the
new channel.
    In the dialplan, all references to these variables refer to the same variable,
regardless of having a prefix or not. Note that setting any version of the
variable removes any other version of the variable, regardless of prefix.

3.4.1 Example
  Set(__FOO=bar) ; Sets an inherited version of "FOO" variable
  Set(FOO=bar) ; Removes the inherited version and sets a local
                  ; variable.
    However, NoOp(${ FOO}) is identical to NoOp(${FOO})


3.5 Selecting Characters from Variables
The format for selecting characters from a variable can be expressed as:
     ${variable_name[:offset[:length]]}
If you want to select the first N characters from the string assigned to a
variable, simply append a colon and the number of characters to skip from
the beginning of the string to the variable name.
     ; Remove the first character of extension, save in "number" variable
     exten => _9X.,1,Set(number=${EXTEN:1})

Assuming we've dialed 918005551234, the value saved to the 'number' vari-
able would be 18005551234. This is useful in situations when we require
users to dial a number to access an outside line, but do not wish to pass the
first digit.
    If you use a negative offset number, Asterisk starts counting from the
end of the string and then selects everything after the new position. The
following example will save the numbers 1234 to the 'number' variable, still
assuming we've dialed 918005551234.
     ; Remove everything before the last four digits of the dialed string
     exten => _9X.,1,Set(number=${EXTEN:-4})
We can also limit the number of characters from our offset position that we
wish to use. This is done by appending a second colon and length value to
the variable name. The following example will save the numbers 555 to the
'number' variable.

                                             42


     ; Only save the middle numbers 555 from the string 918005551234
     exten => _9X.,1,Set(number=${EXTEN:5:3})
The length value can also be used in conjunction with a negative offset.
This may be useful if the length of the string is unknown, but the trailing
digits are. The following example will save the numbers 555 to the 'number'
variable, even if the string starts with more characters than expected (unlike
the previous example).
     ; Save the numbers 555 to the 'number' variable
     exten => _9X.,1,Set(number=${EXTEN:-7:3})
If a negative length value is entered, Asterisk will remove that many charac-
ters from the end of the string.
     ; Set pin to everything but the trailing #.
     exten => _XXXX#,1,Set(pin=${EXTEN:0:-1})


3.6 Expressions
Everything contained inside a bracket pair prefixed by a $ (like $[this]) is
considered as an expression and it is evaluated. Evaluation works similar
to (but is done on a later stage than) variable substitution: the expression
(including the square brackets) is replaced by the result of the expression
evaluation.
    For example, after the sequence:
   exten => 1,1,Set(lala=$[1 + 2])
   exten => 1,2,Set(koko=$[2 * ${lala}])
the value of variable koko is "6".
    and, further:
   exten => 1,1,Set,(lala=$[ 1 + 2 ]);

will parse as intended. Extra spaces are ignored.

3.6.1 Spaces Inside Variables Values
If the variable being evaluated contains spaces, there can be problems.
    For these cases, double quotes around text that may contain spaces will
force the surrounded text to be evaluated as a single token. The double
quotes will be counted as part of that lexical token.
    As an example:

                                              43


  exten => s,6,GotoIf($[ "${CALLERID(name)}" : "Privacy Manager" ]?callerid-liar,s,1:s,7)
   The variable CALLERID(name) could evaluate to "DELOREAN MO-
TORS" (with a space) but the above will evaluate to:

"DELOREAN MOTORS" : "Privacy Manager"
   and will evaluate to 0.
   The above without double quotes would have evaluated to:

DELOREAN MOTORS : Privacy Manager
   and will result in syntax errors, because token DELOREAN is immedi-
ately followed by token MOTORS and the expression parser will not know
how to evaluate this expression, because it does not match its grammar.

3.6.2 Operators
Operators are listed below in order of increasing precedence. Operators with
equal precedence are grouped within { } symbols.

   * expr1 | expr2
      Return the evaluation of expr1 if it is neither an empty string nor zero;
      otherwise, returns the evaluation of expr2.
   * expr1 & expr2
      Return the evaluation of expr1 if neither expression evaluates to an
      empty string or zero; otherwise, returns zero.

   * expr1 {=, >, >=, <, <=, !=} expr2
      Return the results of floating point comparison if both arguments are
      numbers; otherwise, returns the results of string comparison using the
      locale-specific collation sequence. The result of each comparison is 1 if
      the specified relation is true, or 0 if the relation is false.
   * expr1 {+, -} expr2
      Return the results of addition or subtraction of floating point-valued
      arguments.

                                            44


* expr1 {*, /, %} expr2
  Return the results of multiplication, floating point division, or remain-
  der of arguments.
* - expr1
  Return the result of subtracting expr1 from 0. This, the unary minus
  operator, is right associative, and has the same precedence as the !
  operator.

* ! expr1
  Return the result of a logical complement of expr1. In other words, if
  expr1 is null, 0, an empty string, or the string "0", return a 1. Oth-
  erwise, return a 0. It has the same precedence as the unary minus
  operator, and is also right associative.
* expr1 : expr2
  The ':' operator matches expr1 against expr2, which must be a regular
  expression. The regular expression is anchored to the beginning of the
  string with an implicit '^'.
  If the match succeeds and the pattern contains at least one regular
  expression subexpression '...', the string corresponing to '\1' is re-
  turned; otherwise the matching operator returns the number of char-
  acters matched. If the match fails and the pattern contains a regular
  expression subexpression the null string is returned; otherwise 0.
  Normally, the double quotes wrapping a string are left as part of the
  string. This is disastrous to the : operator. Therefore, before the
  regex match is made, beginning and ending double quote characters
  are stripped from both the pattern and the string.

* expr1 =~ expr2
  Exactly the same as the ':' operator, except that the match is not an-
  chored to the beginning of the string. Pardon any similarity to seem-
  ingly similar operators in other programming languages! The ":" and
  "=~" operators share the same precedence.
* expr1 ? expr2 :: expr3


                                        45


      Traditional Conditional operator. If expr1 is a number that evaluates
      to 0 (false), expr3 is result of the this expression evaluation. Otherwise,
      expr2 is the result. If expr1 is a string, and evaluates to an empty string,
      or the two characters (""), then expr3 is the result. Otherwise, expr2 is
      the result. In Asterisk, all 3 exprs will be "evaluated"; if expr1 is "true",
      expr2 will be the result of the "evaluation" of this expression. expr3
      will be the result otherwise. This operator has the lowest precedence.
   * expr1 ~~ expr2
      Concatenation operator. The two exprs are evaluated and turned into
      strings, stripped of surrounding double quotes, and are turned into a
      single string with no invtervening spaces. This operator is new to trunk
      after 1.6.0; it is not needed in existing extensions.conf code. Because of
      the way asterisk evaluates and[ ] constructs (recursively, bottom- up),
      no []or is ever present when the contents of a or[] is evaluated. Thus,
      tokens are usually already merged at evaluation time. But, in AEL,
      various exprs are evaluated raw, and and[] are gathered and treated
      as tokens. And in AEL, no two tokens can sit side by side without
      an intervening operator. So, in AEL, concatenation must be explic-
      itly specified in expressions. This new operator will play well into
      future plans, where expressions ([]constructs,andvariablereferences(
      constructs) are merged into a single grammar.

   Parentheses are used for grouping in the usual manner.
   Operator precedence is applied as one would expect in any of the C or C
derived languages.

3.6.3 Floating Point Numbers
In 1.6 and above, we shifted the $[...] expressions to be calculated via float-
ing point numbers instead of integers. We use 'long double' numbers when
possible, which provide around 16 digits of precision with 12 byte numbers.
   To specify a floating point constant, the number has to have this format:
D.D, where D is a string of base 10 digits. So, you can say 0.10, but you
can't say .10 or 20.-- we hope this is not an excessive restriction!
   Floating point numbers are turned into strings via the '%g'/'%Lg' format
of the printf function set. This allows numbers to still 'look' like integers to
those counting on integer behavior. If you were counting on 1/4 evaluating to

                                              46


0, you need to now say TRUNC(1/4). For a list of all the truncation/rounding
capabilities, see the next section.

3.6.4 Functions
In 1.6 and above, we upgraded the $[] expressions to handle floating point
numbers. Because of this, folks counting on integer behavior would be dis-
rupted. To make the same results possible, some rounding and integer trun-
cation functions have been added to the core of the Expr2 parser. Indeed,
dialplan functions can be called from $[..] expressions without the ${...} op-
erators. The only trouble might be in the fact that the arguments to these
functions must be specified with a comma. If you try to call the MATH
function, for example, and try to say 3 + MATH(7*8), the expression parser
will evaluate 7*8 for you into 56, and the MATH function will most likely
complain that its input doesn't make any sense.
    We also provide access to most of the floating point functions in the C
library. (but not all of them).
    While we don't expect someone to want to do Fourier analysis in the
dialplan, we don't want to preclude it, either.
    Here is a list of the 'builtin' functions in Expr2. All other dialplan func-
tions are available by simply calling them (read-only). In other words, you
don't need to surround function calls in $[...] expressions with ${...}. Don't
jump to conclusions, though! -- you still need to wrap variable names in curly
braces!
   1. COS(x) x is in radians. Results vary from -1 to 1.

   2. SIN(x) x is in radians. Results vary from -1 to 1.
   3. TAN(x) x is in radians.
   4. ACOS(x) x should be a value between -1 and 1.

   5. ASIN(x) x should be a value between -1 and 1.
   6. ATAN(x) returns the arc tangent in radians; between -PI/2 and PI/2.
   7. ATAN2(x,y) returns a result resembling y/x, except that the signs of
      both args are used to determine the quadrant of the result. Its result
      is in radians, between -PI and PI.

                                             47


 8. POW(x,y) returns the value of x raised to the power of y.
 9. SQRT(x) returns the square root of x.

10. FLOOR(x) rounds x down to the nearest integer.
11. CEIL(x) rounds x up to the nearest integer.

12. ROUND(x) rounds x to the nearest integer, but round halfway cases
     away from zero.
13. RINT(x) rounds x to the nearest integer, rounding halfway cases to the
     nearest even integer.
14. TRUNC(x) rounds x to the nearest integer not larger in absolute value.

15. REMAINDER(x,y) computes the remainder of dividing x by y. The
     return value is x - n*y, where n is the value x/y, rounded to the nearest
     integer. If this quotient is 1/2, it is rounded to the nearest even number.
16. EXP(x) returns e to the x power.
17. EXP2(x) returns 2 to the x power.

18. LOG(x) returns the natural logarithm of x.
19. LOG2(x) returns the base 2 log of x.

20. LOG10(x) returns the base 10 log of x.

3.6.5 Examples
 "One Thousand Five Hundred" =~ "(T[^ ]+)"
   returns: Thousand
 "One Thousand Five Hundred" =~ "T[^ ]+"
   returns: 8
 "One Thousand Five Hundred" : "T[^ ]+"
   returns: 0
 "8015551212" : "(...)"
   returns: 801
 "3075551212":"...(...)"
   returns: 555

                                           48


! "One Thousand Five Hundred" =~ "T[^ ]+"
  returns: 0 (because it applies to the string, which is non-null,
                   which it turns to "0", and then looks for the pattern
                   in the "0", and doesn't find it)
!( "One Thousand Five Hundred" : "T[^ ]+" )
  returns: 1 (because the string doesn't start with a word starting
                    with T, so the match evals to 0, and the ! operator
                    inverts it to 1 ).

2 + 8 / 2
  returns 6. (because of operator precedence; the division is done first, then the addition).

2+8/2
  returns 6. Spaces aren't necessary.
(2+8)/2
  returns 5, of course.

(3+8)/2
  returns 5.5 now.
TRUNC((3+8)/2)
  returns 5.

FLOOR(2.5)
  returns 2
FLOOR(-2.5)
  returns -3

CEIL(2.5)
  returns 3.
CEIL(-2.5)
  returns -2.

ROUND(2.5)
  returns 3.
ROUND(3.5)
  returns 4.

ROUND(-2.5)
  returns -3
RINT(2.5)
  returns 2.

RINT(3.5)
  returns 4.

RINT(-2.5)
  returns -2.
RINT(-3.5)
  returns -4.


                                        49


  TRUNC(2.5)
     returns 2.
  TRUNC(3.5)
     returns 3.
  TRUNC(-3.5)
     returns -3.

    Of course, all of the above examples use constants, but would work the
same if any of the numeric or string constants were replaced with a variable
reference ${CALLERID(num)}, for instance.

3.6.6 Numbers Vs. Strings
Tokens consisting only of numbers are converted to 'long double' if possible,
which are from 80 bits to 128 bits depending on the OS, compiler, and
hardware. This means that overflows can occur when the numbers get above
18 digits (depending on the number of bits involved). Warnings will appear
in the logs in this case.

3.6.7 Conditionals
There is one conditional application - the conditional goto :
    exten => 1,2,GotoIf(condition?label1:label2)

    If condition is true go to label1, else go to label2. Labels are interpreted
exactly as in the normal goto command.
    "condition" is just a string. If the string is empty or "0", the condition
is considered to be false, if it's anything else, the condition is true. This is
designed to be used together with the expression syntax described above, eg
:
    exten => 1,2,GotoIf($[${CALLERID(all)} = 123456]?2,1:3,1)
    Example of use :
  exten => s,2,Set(vara=1)
  exten => s,3,Set(varb=$[${vara} + 2])
  exten => s,4,Set(varc=$[${varb} * 2])
  exten => s,5,GotoIf($[${varc} = 6]?99,1:s,6)


                                             50


3.6.8 Parse Errors
Syntax errors are now output with 3 lines.
    If the extensions.conf file contains a line like:

  exten => s,6,GotoIf($[ "${CALLERID(num)}" = "3071234567" & & "${CALLERID(name)}" : "Privacy Manager" ]?callerid-liar,s,1:s,7)
    You may see an error in /var/log/asterisk/messages like this:
  Jul 15 21:27:49 WARNING[1251240752]: ast_yyerror(): syntax error: parse error, unexpected TOK_AND, expecting TOK_MINUS or TOK_LP or TOKEN; Input:
  "3072312154" = "3071234567" & & "Steves Extension" : "Privacy Manager"
                  ^

    The log line tells you that a syntax error was encountered. It now also
tells you (in grand standard bison format) that it hit an "AND" (&) token
unexpectedly, and that was hoping for for a MINUS (-), LP (left parenthesis),
or a plain token (a string or number).
    The next line shows the evaluated expression, and the line after that, the
position of the parser in the expression when it became confused, marked
with the "^" character.

3.6.9 NULL Strings
Testing to see if a string is null can be done in one of two different ways:
    exten => _XX.,1,GotoIf($["${calledid}" != ""]?3)
       or
    exten => _XX.,1,GotoIf($[foo${calledid} != foo]?3)
    The second example above is the way suggested by the WIKI. It will work
as long as there are no spaces in the evaluated value.
    The first way should work in all cases, and indeed, might now be the
safest way to handle this situation.

3.6.10 Warning

If you need to do complicated things with strings, asterisk expressions is
most likely NOT the best way to go about it. AGI scripts are an excellent
option to this need, and make available the full power of whatever language
you desire, be it Perl, C, C++, Cobol, RPG, Java, Snobol, PL/I, Scheme,
Common Lisp, Shell scripts, Tcl, Forth, Modula, Pascal, APL, assembler,
etc.

                                              51


3.6.11 Incompatabilities
The asterisk expression parser has undergone some evolution. It is hoped
that the changes will be viewed as positive.
   The "original" expression parser had a simple, hand-written scanner, and
a simple bison grammar. This was upgraded to a more involved bison gram-
mar, and a hand-written scanner upgraded to allow extra spaces, and to
generate better error diagnostics. This upgrade required bison 1.85, and
part of the user community felt the pain of having to upgrade their bison
version.
   The next upgrade included new bison and flex input files, and the makefile
was upgraded to detect current version of both flex and bison, conditionally
compiling and linking the new files if the versions of flex and bison would
allow it.
   If you have not touched your extensions.conf files in a year or so, the above
upgrades may cause you some heartburn in certain circumstances, as several
changes have been made, and these will affect asterisk's behavior on legacy
extension.conf constructs. The changes have been engineered to minimize
these conflicts, but there are bound to be problems.
   The following list gives some (and most likely, not all) of areas of possible
concern with "legacy" extension.conf files:

  1. Tokens separated by space(s). Previously, tokens were separated by
      spaces. Thus, ' 1 + 1 ' would evaluate to the value '2', but '1+1' would
      evaluate to the string '1+1'. If this behavior was depended on, then
      the expression evaluation will break. '1+1' will now evaluate to '2',
      and something is not going to work right. To keep such strings from
      being evaluated, simply wrap them in double quotes: ' "1+1" '
  2. The colon operator. In versions previous to double quoting, the colon
      operator takes the right hand string, and using it as a regex pattern,
      looks for it in the left hand string. It is given an implicit ^operator at the
      beginning, meaning the pattern will match only at the beginning of the
      left hand string. If the pattern or the matching string had double quotes
      around them, these could get in the way of the pattern match. Now,
      the wrapping double quotes are stripped from both the pattern and the
      left hand string before applying the pattern. This was done because
      it recognized that the new way of scanning the expression doesn't use
      spaces to separate tokens, and the average regex expression is full of

                                               52


      operators that the scanner will recognize as expression operators. Thus,
      unless the pattern is wrapped in double quotes, there will be trouble.
      For instance, ${VAR1} : (Who|What*)+ may have have worked before,
      but unless you wrap the pattern in double quotes now, look out for
      trouble! This is better: "${VAR1}" : "(Who|What*)+" and should
      work as previous.
   3. Variables and Double Quotes Before these changes, if a variable's value
      contained one or more double quotes, it was no reason for concern. It
      is now!

   4. LE, GE, NE operators removed. The code supported these operators,
      but they were not documented. The symbolic operators, <=, >=, and
      != should be used instead.
   5. Added the unary '-' operator. So you can 3+ -4 and get -1.
   6. Added the unary '!' operator, which is a logical complement. Basically,
      if the string or number is null, empty, or '0', a '1' is returned. Otherwise
      a '0' is returned.
   7. Added the '=~' operator, just in case someone is just looking for match
      anywhere in the string. The only diff with the ':' is that match doesn't
      have to be anchored to the beginning of the string.

   8. Added the conditional operator 'expr1 ? true expr :: false expr' First,
      all 3 exprs are evaluated, and if expr1 is false, the 'false expr' is returned
      as the result. See above for details.
   9. Unary operators '-' and '!' were made right associative.

3.6.12 Debugging Hints
There are two utilities you can build to help debug the $[ ] in your exten-
sions.conf file.
   The first, and most simplistic, is to issue the command:
  make testexpr2
in the top level asterisk source directory. This will build a small executable,
that is able to take the first command line argument, and run it thru the

                                               53


expression parser. No variable substitutions will be performed. It might be
safest to wrap the expression in single quotes...
  testexpr2 '2*2+2/2'
is an example.
    And, in the utils directory, you can say:
  make check_expr
and a small program will be built, that will check the file mentioned in
the first command line argument, for any expressions that might be have
problems when you move to flex-2.5.31. It was originally designed to help
spot possible incompatibilities when moving from the pre-2.5.31 world to the
upgraded version of the lexer.
    But one more capability has been added to check expr, that might make
it more generally useful. It now does a simple minded evaluation of all
variables, and then passes the $[] exprs to the parser. If there are any parse
errors, they will be reported in the log file. You can use check expr to do
a quick sanity check of the expressions in your extensions.conf file, to see if
they pass a crude syntax check.
    The "simple-minded" variable substitution replaces ${varname} variable
references with '555'. You can override the 555 for variable values, by entering
in var=val arguments after the filename on the command line. So...
   check_expr /etc/asterisk/extensions.conf CALLERID(num)=3075551212 DIALSTATUS=TORTURE EXTEN=121

will substitute any ${CALLERID(num)} variable references with 3075551212,
any ${DIALSTATUS} variable references with 'TORTURE', and any ${EXTEN}
references with '121'. If there is any fancy stuff going on in the reference,
like ${EXTEN:2}, then the override will not work. Everything in the ${...}
has to match. So, to substitute ${EXTEN:2} references, you'd best say:
   check_expr /etc/asterisk/extensions.conf CALLERID(num)=3075551212 DIALSTATUS=TORTURE EXTEN:2=121
on stdout, you will see something like:
   OK -- $[ "${DIALSTATUS}" = "TORTURE" | "${DIALSTATUS}" = "DONTCALL" ] at line 416
    In the expr2 log file that is generated, you will see:
   line 416, evaluation of $[ "TORTURE" = "TORTURE" | "TORTURE" = "DONTCALL" ] result: 1

    check expr is a very simplistic algorithm, and it is far from being guar-
anteed to work in all cases, but it is hoped that it will be useful.

                                              54


3.7 Asterisk standard channel variables
There are a number of variables that are defined or read by Asterisk. Here is
a list of them. More information is available in each application's help text.
All these variables are in UPPER CASE only.
   Variables marked with a * are builtin functions and can't be set, only
read in the dialplan. Writes to such variables are silently ignored.

${CDR(accountcode)} * Account code (if specified)
${BLINDTRANSFER} The name of the channel on the other side of a blind transfer
${BRIDGEPEER} Bridged peer
${BRIDGEPVTCALLID} Bridged peer PVT call ID (SIP Call ID if a SIP call)
${CALLERID(ani)} * Caller ANI (PRI channels)
${CALLERID(ani2)} * ANI2 (Info digits) also called Originating line information or OLI
${CALLERID(all)} * Caller ID
${CALLERID(dnid)} * Dialed Number Identifier
${CALLERID(name)} * Caller ID Name only
${CALLERID(num)} * Caller ID Number only
${CALLERID(rdnis)} * Redirected Dial Number ID Service
${CALLINGANI2} * Caller ANI2 (PRI channels)
${CALLINGPRES} * Caller ID presentation for incoming calls (PRI channels)
${CALLINGTNS} * Transit Network Selector (PRI channels)
${CALLINGTON} * Caller Type of Number (PRI channels)
${CHANNEL} * Current channel name
${CONTEXT} * Current context
${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS
                                 (Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
${DB_RESULT} Result value of DB_EXISTS() dial plan function
${EPOCH} * Current unix style epoch
${EXTEN} * Current extension
${ENV(VAR)} Environmental variable VAR
${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority
                                 after a blind transfer (use ^ characters in place of
                                 | to separate context/extension/priority when setting
                                 this variable from the dialplan)
${HANGUPCAUSE} * Asterisk cause of hangup (inbound/outbound)
${HINT} * Channel hints for this extension
${HINTNAME} * Suggested Caller*ID name for this extension
                                         55


${INVALID_EXTEN} The invalid called extension (used in the "i" extension)
${LANGUAGE} * Current language (Deprecated; use ${LANGUAGE()})
${LEN(VAR)} * String length of VAR (integer)
${PRIORITY} * Current priority in the dialplan
${PRIREDIRECTREASON} Reason for redirect on PRI, if a call was directed
${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS
                                (Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
${TRANSFER_CONTEXT} Context for transferred calls
${FORWARD_CONTEXT} Context for forwarded calls
${UNIQUEID} * Current call unique identifier
${SYSTEMNAME} * value of the systemname option of asterisk.conf
${ENTITYID} * Global Entity ID set automatically, or from asterisk.conf

3.7.1 Application return values
Many applications return the result in a variable that you read to get the
result of the application. These status fields are unique for each application.
For the various status values, see each application's help text.
${AGISTATUS} * agi()
${AQMSTATUS} * addqueuemember()
${AVAILSTATUS} * chanisavail()
${CHECKGROUPSTATUS} * checkgroup()
${CHECKMD5STATUS} * checkmd5()
${CPLAYBACKSTATUS} * controlplayback()
${DIALSTATUS} * dial()
${DBGETSTATUS} * dbget()
${ENUMSTATUS} * enumlookup()
${HASVMSTATUS} * hasnewvoicemail()
${LOOKUPBLSTATUS} * lookupblacklist()
${OSPAUTHSTATUS} * ospauth()
${OSPLOOKUPSTATUS} * osplookup()
${OSPNEXTSTATUS} * ospnext()
${OSPFINISHSTATUS} * ospfinish()
${PARKEDAT} * parkandannounce()
${PLAYBACKSTATUS} * playback()
${PQMSTATUS} * pausequeuemember()
${PRIVACYMGRSTATUS} * privacymanager()

                                       56


${QUEUESTATUS} * queue()
${RQMSTATUS} * removequeuemember()
${SENDIMAGESTATUS} * sendimage()
${SENDTEXTSTATUS} * sendtext()
${SENDURLSTATUS} * sendurl()
${SYSTEMSTATUS} * system()
${TRANSFERSTATUS} * transfer()
${TXTCIDNAMESTATUS} * txtcidname()
${UPQMSTATUS} * unpausequeuemember()
${VMSTATUS} * voicmail()
${VMBOXEXISTSSTATUS} * vmboxexists()
${WAITSTATUS} * waitforsilence()

3.7.2 Various application variables
${CURL} * Resulting page content for curl()
${ENUM} * Result of application EnumLookup
${EXITCONTEXT} Context to exit to in IVR menu (app background())
                           or in the RetryDial() application
${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor())
${MONITOR_EXEC} Application to execute after monitoring a call
${MONITOR_EXEC_ARGS} Arguments to application
${MONITOR_FILENAME} File for monitoring (recording) calls in queue
${QUEUE_PRIO} Queue priority
${QUEUE_MAX_PENALTY} Maximum member penalty allowed to answer caller
${QUEUE_MIN_PENALTY} Minimum member penalty allowed to answer caller
${QUEUESTATUS} Status of the call, one of:
                           (TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL)
${RECORDED_FILE} * Recorded file in record()
${TALK_DETECTED} * Result from talkdetect()
${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record)
${TOUCH_MONITOR_PREF} * The prefix for automonitor recording filenames.
${TOUCH_MONITOR_FORMAT} The audio format to use with Touch Monitor (auto record)
${TOUCH_MONITOR_OUTPUT} * Recorded file from Touch Monitor (auto record)
${TOUCH_MONITOR_MESSAGE_START} Recorded file to play for both channels at start of monitoring session
${TOUCH_MONITOR_MESSAGE_STOP} Recorded file to play for both channels at end of monitoring session
${TXTCIDNAME} * Result of application TXTCIDName
${VPB_GETDTMF} chan_vpb
                                57


3.7.3 The MeetMe Conference Bridge
${MEETME_RECORDINGFILE} Name of file for recording a conference with
                              the "r" option
${MEETME_RECORDINGFORMAT} Format of file to be recorded
${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting
${MEETME_AGI_BACKGROUND} AGI script for Meetme (DAHDI only)
${MEETMESECS} * Number of seconds a user participated in a MeetMe conference
${CONF_LIMIT_TIMEOUT_FILE} File to play when time is up. Used with the L() option.
${CONF_LIMIT_WARNING_FILE} File to play as warning if 'y' is defined.
                              The default is to say the time remaining. Used with the L() option.
${MEETMEBOOKID} * This variable exposes the bookid column for a realtime configured conference bridge.

3.7.4 The VoiceMail() application
${VM_CATEGORY} Sets voicemail category
${VM_NAME} * Full name in voicemail
${VM_DUR} * Voicemail duration
${VM_MSGNUM} * Number of voicemail message in mailbox
${VM_CALLERID} * Voicemail Caller ID (Person leaving vm)
${VM_CIDNAME} * Voicemail Caller ID Name
${VM_CIDNUM} * Voicemail Caller ID Number
${VM_DATE} * Voicemail Date
${VM_MESSAGEFILE} * Path to message left by caller

3.7.5 The VMAuthenticate() application
${AUTH_MAILBOX} * Authenticated mailbox
${AUTH_CONTEXT} * Authenticated mailbox context
3.7.6 DUNDiLookup()

${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()
${DUNDDEST} * The Destination of the result from a call to DUNDiLookup()
3.7.7 chan dahdi
${ANI2} * The ANI2 Code provided by the network on the incoming call.

                               58


                         (ie, Code 29 identifies call as a Prison/Inmate Call)
${CALLTYPE} * Type of call (Speech, Digital, etc)
${CALLEDTON} * Type of number for incoming PRI extension
                         i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific,
                         4=subscriber, 6=abbreviated, 7=reserved
${CALLINGSUBADDR} * Caller's PRI Subaddress
${FAXEXTEN} * The extension called before being redirected to "fax"
${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
${SMDI_VM_TYPE} * When an call is received with an SMDI message, the 'type'
                         of message 'b' or 'u'

3.7.8 chan sip
${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
${SIPFROMDOMAIN} Set SIP domain on outbound calls
${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
${SIP_CODEC} Set the SIP codec for a call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
${RTPAUDIOQOS} RTCP QoS report for the audio of this call
${RTPVIDEOQOS} RTCP QoS report for the video of this call

3.7.9 chan agent
${AGENTMAXLOGINTRIES} Set the maximum number of failed logins
${AGENTUPDATECDR} Whether to update the CDR record with Agent channel data
${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out
${AGENTACKCALL} Whether the agent should acknowledge the incoming call
${AGENTAUTOLOGOFF} Auto logging off for an agent
${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls
${AGENTNUMBER} * Agent number (username) set at login
${AGENTSTATUS} * Status of login ( fail | on | off )
${AGENTEXTEN} * Extension for logged in agent
3.7.10 The Dial() application

${DIALEDPEERNAME} * Dialed peer name
                                59


${DIALEDPEERNUMBER} * Dialed peer number
${DIALEDTIME} * Time for the call (seconds). Is only set if call was answered.
${ANSWEREDTIME} * Time from answer to hangup (seconds)
${DIALSTATUS} * Status of the call, one of:
                          (CHANUNAVAIL | CONGESTION | BUSY | NOANSWER
                          | ANSWER | CANCEL | DONTCALL | TORTURE)
${DYNAMIC_FEATURES} * The list of features (from the [applicationmap] section of
                          features.conf) to activate during the call, with feature
                          names separated by '#' characters
${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits
${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits
${LIMIT_WARNING_FILE} Soundfile for call limits
${LIMIT_TIMEOUT_FILE} Soundfile for call limits
${LIMIT_CONNECT_FILE} Soundfile for call limits
${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup)
                             * See "show application dial" for more information

3.7.11 The chanisavail() application
${AVAILCHAN} * the name of the available channel if one was found
${AVAILORIGCHAN} * the canonical channel name that was used to create the channel
${AVAILSTATUS} * Status of requested channel
3.7.12 Dialplan Macros

${MACRO_EXTEN} * The calling extensions
${MACRO_CONTEXT} * The calling context
${MACRO_PRIORITY} * The calling priority
${MACRO_OFFSET} Offset to add to priority at return from macro
3.7.13 The ChanSpy() application
${SPYGROUP} * A ':' (colon) separated list of group names.
                          (To be set on spied on channel and matched against the g(grp) option)

3.7.14 OSP
${OSPINHANDLE} OSP handle of in_bound call

                                60


${OSPINTIMELIMIT} Duration limit for in_bound call
${OSPOUTHANDLE} OSP handle of out_bound call
${OSPTECH} OSP technology
${OSPDEST} OSP destination
${OSPCALLING} OSP calling number
${OSPOUTTOKEN} OSP token to use for out_bound call
${OSPOUTTIMELIMIT} Duration limit for out_bound call
${OSPRESULTS} Number of remained destinations






















                         61




Chapter 4

AEL: Asterisk Extension Language


4.1 Introduction

AEL is a specialized language intended purely for describing Asterisk dial
plans.
   The current version was written by Steve Murphy, and is a rewrite of the
original version.
   This new version further extends AEL, and provides more flexible syntax,
better error messages, and some missing functionality.
   AEL is really the merger of 4 different 'languages', or syntaxes:
   * The first and most obvious is the AEL syntax itself. A BNF is provided
      near the end of this document.
   * The second syntax is the Expression Syntax, which is normally handled
      by Asterisk extension engine, as expressions enclosed in $[...]. The right
      hand side of assignments are wrapped in $[ ... ] by AEL, and so are
      the if and while expressions, among others.
   * The third syntax is the Variable Reference Syntax, the stuff enclosed in
      ${..} curly braces. It's a bit more involved than just putting a variable
      name in there. You can include one of dozens of 'functions', and their
      arguments, and there are even some string manipulation notation in
      there.
   * The last syntax that underlies AEL, and is not used directly in AEL,
      is the Extension Language Syntax. The extension language is what

                                            62


       you see in extensions.conf, and AEL compiles the higher level AEL
       language into extensions and priorities, and passes them via function
       calls into Asterisk. Embedded in this language is the Application/AGI
       commands, of which one application call per step, or priority can be
       made. You can think of this as a "macro assembler" language, that
       AEL will compile into.
    Any programmer of AEL should be familiar with its syntax, of course, as
well as the Expression syntax, and the Variable syntax.

4.2 Asterisk in a Nutshell

Asterisk acts as a server. Devices involved in telephony, like DAHDI cards,
or Voip phones, all indicate some context that should be activated in their
behalf. See the config file formats for IAX, SIP, dahdi.conf, etc. They all help
describe a device, and they all specify a context to activate when somebody
picks up a phone, or a call comes in from the phone company, or a voip
phone, etc.

4.2.1 Contexts
Contexts are a grouping of extensions.
    Contexts can also include other contexts. Think of it as a sort of merge
operation at runtime, whereby the included context's extensions are added
to the contexts making the inclusion.

4.2.2 Extensions and priorities
A Context contains zero or more Extensions. There are several predefined
extensions. The "s" extension is the "start" extension, and when a device
activates a context the "s" extension is the one that is going to be run.
Other extensions are the timeout "t" extension, the invalid response, or "i"
extension, and there's a "fax" extension. For instance, a normal call will
activate the "s" extension, but an incoming FAX call will come into the
"fax" extension, if it exists. (BTW, asterisk can tell it's a fax call by the
little "beep" that the calling fax machine emits every so many seconds.).
    Extensions contain several priorities, which are individual instructions to
perform. Some are as simple as setting a variable to a value. Others are as
                                             63


complex as initiating the Voicemail application, for instance. Priorities are
executed in order.
   When the 's" extension completes, asterisk waits until the timeout for
a response. If the response matches an extension's pattern in the context,
then control is transferred to that extension. Usually the responses are tones
emitted when a user presses a button on their phone. For instance, a context
associated with a desk phone might not have any "s" extension. It just plays
a dialtone until someone starts hitting numbers on the keypad, gather the
number, find a matching extension, and begin executing it. That extension
might Dial out over a connected telephone line for the user, and then connect
the two lines together.
   The extensions can also contain "goto" or "jump" commands to skip to
extensions in other contexts. Conditionals provide the ability to react to
different stimuli, and there you have it.

4.2.3 Macros
Think of a macro as a combination of a context with one nameless extension,
and a subroutine. It has arguments like a subroutine might. A macro call
can be made within an extension, and the individual statements there are
executed until it ends. At this point, execution returns to the next statement
after the macro call. Macros can call other macros. And they work just like
function calls.

4.2.4 Applications
Application calls, like "Dial()", or "Hangup()", or "Answer()", are available
for users to use to accomplish the work of the dialplan. There are over 145
of them at the moment this was written, and the list grows as new needs
and wants are uncovered. Some applications do fairly simple things, some
provide amazingly complex services.
   Hopefully, the above objects will allow you do anything you need to in
the Asterisk environment!





                                          64


4.3 Getting Started

The AEL parser (res ael.so) is completely separate from the module that
parses extensions.conf (pbx config.so). To use AEL, the only thing that has
to be done is the module res ael.so must be loaded by Asterisk. This will
be done automatically if using 'autoload=yes' in /etc/asterisk/modules.
conf. When the module is loaded, it will look for 'extensions.ael' in /etc/
asterisk/. extensions.conf and extensions.ael can be used in conjunction
with each other if that is what is desired. Some users may want to keep
extensions.conf for the features that are configured in the 'general' section of
extensions.conf.
    To reload extensions.ael, the following command can be issued at the CLI:
    *CLI> ael reload

4.4 Debugging
Right at this moment, the following commands are available, but do nothing:
    Enable AEL contexts debug
    *CLI> ael debug contexts
    Enable AEL macros debug
    *CLI> ael debug macros
    Enable AEL read debug
    *CLI> ael debug read
    Enable AEL tokens debug
    *CLI> ael debug tokens
    Disable AEL debug messages
    *CLI> ael no debug
    If things are going wrong in your dialplan, you can use the following
facilities to debug your file:
    1. The messages log in /var/log/asterisk. (from the checks done at
load time). 2. the "show dialplan" command in asterisk 3. the standalone
executable, "aelparse" built in the utils/ dir in the source.

4.5 About "aelparse"
You can use the "aelparse" program to check your extensions.ael file before
feeding it to asterisk. Wouldn't it be nice to eliminate most errors before

                                            65


giving the file to asterisk?
    aelparse is compiled in the utils directory of the asterisk release. It isn't
installed anywhere (yet). You can copy it to your favorite spot in your PATH.
    aelparse has two optional arguments:
    * -d

          -- Override the normal location of the config file dir, (usually /etc/
             asterisk), and use the current directory instead as the config file
             dir. Aelparse will then expect to find the file "./extensions.ael" in
             the current directory, and any included files in the current direc-
             tory as well.
    * -n
          -- don't show all the function calls to set priorities and contexts
             within asterisk. It will just show the errors and warnings from the
             parsing and semantic checking phases.


4.6 General Notes about Syntax
Note that the syntax and style are now a little more free-form. The opening
'' (curly-braces) do not have to be on the same line as the keyword that
precedes them. Statements can be split across lines, as long as tokens are
not broken by doing so. More than one statement can be included on a single
line. Whatever you think is best!
    You can just as easily say,
  if(${x}=1) { NoOp(hello!); goto s,3; } else { NoOp(Goodbye!); goto s,12; }

as you can say:
  if(${x}=1)
  {
         NoOp(hello!);
     goto s,3;
  }
  else
  {
         NoOp(Goodbye!);
         goto s,12;
  }

                                             66


    or:
  if(${x}=1) {
         NoOp(hello!);
     goto s,3;
  } else {
         NoOp(Goodbye!);
         goto s,12;
  }

    or:
  if (${x}=1) {
         NoOp(hello!); goto s,3;
  } else {
         NoOp(Goodbye!); goto s,12;
  }

4.7 Keywords
The AEL keywords are case-sensitive. If an application name and a keyword
overlap, there is probably good reason, and you should consider replacing
the application call with an AEL statement. If you do not wish to do so,
you can still use the application, by using a capitalized letter somewhere in
its name. In the Asterisk extension language, application names are NOT
case-sensitive.
    The following are keywords in the AEL language:

    * abstract
    * context

    * macro
    * globals
    * ignorepat

    * switch
    * if
    * ifTime

    * else
                                           67


* random
* goto

* jump
* local
* return

* break
* continue
* regexten

* hint
* for
* while

* case
* pattern
* default NOTE: the "default" keyword can be used as a context name,
  for those who would like to do so.

* catch
* switches
* eswitches
* includes







                                 68


4.8 Procedural Interface and Internals

AEL first parses the extensions.ael file into a memory structure representing
the file. The entire file is represented by a tree of "pval" structures linked
together.
    This tree is then handed to the semantic check routine.
    Then the tree is handed to the compiler.
    After that, it is freed from memory.
    A program could be written that could build a tree of pval structures,
and a pretty printing function is provided, that would dump the data to a
file, or the tree could be handed to the compiler to merge the data into the
asterisk dialplan. The modularity of the design offers several opportunities
for developers to simplify apps to generate dialplan data.

4.8.1 AEL version 2 BNF
(hopefully, something close to bnf).
    First, some basic objects

   ------------------------
   <word> a lexical token consisting of characters matching this pattern: [-a-zA-Z0-9"_/.\<\>\*\+!$#\[\]][-a-zA-Z0-9"_/.!\*\+\<\>\{\}$#\[\]]*
   <word3-list> a concatenation of up to 3 <word>s.
   <collected-word> all characters encountered until the character that follows the <collected-word> in the grammar.
   -------------------------
   <file> :== <objects>

   <objects> :== <object>
              | <objects> <object>
   <object> :== <context>
            | <macro>
            | <globals>
            | ';'

   <context> :== 'context' <word> '{' <elements> '}'
               | 'context' <word> '{' '}'
               | 'context' 'default' '{' <elements> '}'
               | 'context' 'default' '{' '}'
               | 'abstract' 'context' <word> '{' <elements> '}'
               | 'abstract' 'context' <word> '{' '}'
               | 'abstract' 'context' 'default' '{' <elements> '}'
               | 'abstract' 'context' 'default' '{' '}'


                                               69


<macro> :== 'macro' <word> '(' <arglist> ')' '{' <macro_statements> '}'
       | 'macro' <word> '(' <arglist> ')' '{' '}'
       | 'macro' <word> '(' ')' '{' <macro_statements> '}'
       | 'macro' <word> '(' ')' '{' '}'

<globals> :== 'globals' '{' <global_statements> '}'
         | 'globals' '{' '}'

<global_statements> :== <global_statement>
                   | <global_statements> <global_statement>


<global_statement> :== <word> '=' <collected-word> ';'

<arglist> :== <word>
         | <arglist> ',' <word>

<elements> :== <element>
             | <elements> <element>

<element> :== <extension>
         | <includes>
         | <switches>
         | <eswitches>
         | <ignorepat>
         | <word> '=' <collected-word> ';'
         | 'local' <word> '=' <collected-word> ';'
         | ';'


<ignorepat> :== 'ignorepat' '=>' <word> ';'

<extension> :== <word> '=>' <statement>
           | 'regexten' <word> '=>' <statement>
           | 'hint' '(' <word3-list> ')' <word> '=>' <statement>
           | 'regexten' 'hint' '(' <word3-list> ')' <word> '=>' <statement>

<statements> :== <statement>
            | <statements> <statement>

<if_head> :== 'if' '(' <collected-word> ')'
<random_head> :== 'random' '(' <collected-word> ')'

<ifTime_head> :== 'ifTime' '(' <word3-list> ':' <word3-list> ':' <word3-list> '|' <word3-list> '|' <word3-list> '|' <word3-list> ')'
                       | 'ifTime' '(' <word> '|' <word3-list> '|' <word3-list> '|' <word3-list> ')'

<word3-list> :== <word>


                                         70


       | <word> <word>
       | <word> <word> <word>
<switch_head> :== 'switch' '(' <collected-word> ')' '{'


<statement> :== '{' <statements> '}'
       | <word> '=' <collected-word> ';'
       | 'local' <word> '=' <collected-word> ';'
       | 'goto' <target> ';'
       | 'jump' <jumptarget> ';'
       | <word> ':'
       | 'for' '(' <collected-word> ';' <collected-word> ';' <collected-word> ')' <statement>
       | 'while' '(' <collected-word> ')' <statement>
       | <switch_head> '}'
       | <switch_head> <case_statements> '}'
       | '&' macro_call ';'
       | <application_call> ';'
       | <application_call> '=' <collected-word> ';'
       | 'break' ';'
       | 'return' ';'
       | 'continue' ';'
       | <random_head> <statement>
       | <random_head> <statement> 'else' <statement>
       | <if_head> <statement>
       | <if_head> <statement> 'else' <statement>
       | <ifTime_head> <statement>
       | <ifTime_head> <statement> 'else' <statement>
       | ';'
<target> :== <word>
       | <word> '|' <word>
       | <word> '|' <word> '|' <word>
       | 'default' '|' <word> '|' <word>
       | <word> ',' <word>
       | <word> ',' <word> ',' <word>
       | 'default' ',' <word> ',' <word>

<jumptarget> :== <word>
               | <word> ',' <word>
               | <word> ',' <word> '@' <word>
               | <word> '@' <word>
               | <word> ',' <word> '@' 'default'
               | <word> '@' 'default'
<macro_call> :== <word> '(' <eval_arglist> ')'
       | <word> '(' ')'

<application_call_head> :== <word> '('
<application_call> :== <application_call_head> <eval_arglist> ')'
       | <application_call_head> ')'

<eval_arglist> :== <collected-word>
       | <eval_arglist> ',' <collected-word>
       | /* nothing */
       | <eval_arglist> ',' /* nothing */


                                         71


  <case_statements> :== <case_statement>
         | <case_statements> <case_statement>

  <case_statement> :== 'case' <word> ':' <statements>
         | 'default' ':' <statements>
         | 'pattern' <word> ':' <statements>
         | 'case' <word> ':'
         | 'default' ':'
         | 'pattern' <word> ':'
  <macro_statements> :== <macro_statement>
         | <macro_statements> <macro_statement>

  <macro_statement> :== <statement>
         | 'catch' <word> '{' <statements> '}'
  <switches> :== 'switches' '{' <switchlist> '}'
         | 'switches' '{' '}'
  <eswitches> :== 'eswitches' '{' <switchlist> '}'
         | 'eswitches' '{' '}'

  <switchlist> :== <word> ';'
         | <switchlist> <word> ';'
  <includeslist> :== <includedname> ';'
         | <includedname> '|' <word3-list> ':' <word3-list> ':' <word3-list> '|' <word3-list> '|' <word3-list> '|' <word3-list> ';'
         | <includedname> '|' <word> '|' <word3-list> '|' <word3-list> '|' <word3-list> ';'
         | <includeslist> <includedname> ';'
         | <includeslist> <includedname> '|' <word3-list> ':' <word3-list> ':' <word3-list> '|' <word3-list> '|' <word3-list> '|' <word3-list> ';'
         | <includeslist> <includedname> '|' <word> '|' <word3-list> '|' <word3-list> '|' <word3-list> ';'
  <includedname> :== <word>
          | 'default'

  <includes> :== 'includes' '{' <includeslist> '}'
         | 'includes' '{' '}'

4.9 AEL Example USAGE

4.9.1 Comments

Comments begin with // and end with the end of the line.
   Comments are removed by the lexical scanner, and will not be recognized
in places where it is busy gathering expressions to wrap in $[] , or inside
application call argument lists. The safest place to put comments is after
terminating semicolons, or on otherwise empty lines.



                                            72


4.9.2 Context
Contexts in AEL represent a set of extensions in the same way that they do
in extensions.conf.
  context default {

  }
    A context can be declared to be "abstract", in which case, this declaration
expresses the intent of the writer, that this context will only be included by
another context, and not "stand on its own". The current effect of this
keyword is to prevent "goto " statements from being checked.
  abstract context longdist {
       _1NXXNXXXXXX => NoOp(generic long distance dialing actions in the US);
  }

4.9.3 Extensions
To specify an extension in a context, the following syntax is used. If more
than one application is be called in an extension, they can be listed in order
inside of a block.
  context default {
      1234 => Playback(tt-monkeys);
      8000 => {
           NoOp(one);
           NoOp(two);
           NoOp(three);
      };
      _5XXX => NoOp(it's a pattern!);
  }

    Two optional items have been added to the AEL syntax, that allow the
specification of hints, and a keyword, regexten, that will force the numbering
of priorities to start at 2.
    The ability to make extensions match by CID is preserved in AEL; just
use '/' and the CID number in the specification. See below.
  context default {
      regexten _5XXX => NoOp(it's a pattern!);
  }
  context default {
      hint(Sip/1) _5XXX => NoOp(it's a pattern!);
  }

                                             73


  context default {
      regexten hint(Sip/1) _5XXX => NoOp(it's a pattern!);
  }
   The regexten must come before the hint if they are both present.
   CID matching is done as with the extensions.conf file. Follow the exten-
sion name/number with a slash (/) and the number to match against the
Caller ID:

  context zoombo
  {
    819/7079953345 => { NoOp(hello, 3345); }
  }
   In the above, the 819/7079953345 extension will only be matched if the
CallerID is 7079953345, and the dialed number is 819. Hopefully you have
another 819 extension defined for all those who wish 819, that are not so
lucky as to have 7079953345 as their CallerID!

4.9.4 Includes
Contexts can be included in other contexts. All included contexts are listed
within a single block.
  context default {
      includes {
           local;
           longdistance;
           international;
      }
  }

   Time-limited inclusions can be specified, as in extensions.conf format,
with the fields described in the wiki page Asterisk cmd GotoIfTime.
  context default {
      includes {
           local;
           longdistance|16:00-23:59|mon-fri|*|*;
           international;
      }
  }




                                            74


4.9.5 #include
You can include other files with the #include "filepath" construct.

     #include "/etc/asterisk/testfor.ael"
   An interesting property of the #include, is that you can use it almost
anywhere in the .ael file. It is possible to include the contents of a file in a
macro, context, or even extension. The #include does not have to occur at
the beginning of a line. Included files can include other files, up to 50 levels
deep. If the path provided in quotes is a relative path, the parser looks in
the config file directory for the file (usually /etc/asterisk).

4.9.6 Dialplan Switches
Switches are listed in their own block within a context. For clues as to
what these are used for, see Asterisk - dual servers, and Asterisk config
extensions.conf.
  context default {
      switches {
           DUNDi/e164;
           IAX2/box5;
      };
      eswitches {
           IAX2/context@${CURSERVER};
      }
  }

4.9.7 Ignorepat

ignorepat can be used to instruct channel drivers to not cancel dialtone upon
receipt of a particular pattern. The most commonly used example is '9'.
  context outgoing {
      ignorepat => 9;
  }
4.9.8 Variables

Variables in Asterisk do not have a type, so to define a variable, it just has
to be specified with a value.
   Global variables are set in their own block.

                                             75


  globals {
      CONSOLE=Console/dsp;
      TRUNK=DAHDI/g2;
  }
    Variables can be set within extensions as well.
  context foo {
      555 => {
           x=5;
           y=blah;
           divexample=10/2
           NoOp(x is ${x} and y is ${y} !);
      }
  }
    NOTE: AEL wraps the right hand side of an assignment with $[ ] to
allow expressions to be used If this is unwanted, you can protect the right
hand side from being wrapped by using the Set() application. Read the
README.variables about the requirements and behavior of $[ ] expressions.
    NOTE: These things are wrapped up in a $[ ] expression: The while()
test; the if() test; the middle expression in the for( x; y; z) statement (the y
expression); Assignments - the right hand side, so a = b -> Set(a=$[b])
    Writing to a dialplan function is treated the same as writing to a variable.
  context blah {
      s => {
           CALLERID(name)=ChickenMan;
           NoOp(My name is ${CALLERID(name)} !);
      }
  }
    You can declare variables in Macros, as so:
  Macro myroutine(firstarg, secondarg)
  {
    Myvar=1;
    NoOp(Myvar is set to ${myvar});
  }

4.9.9 Local Variables
In 1.2, and 1.4, ALL VARIABLES are CHANNEL variables, including the
function arguments and associated ARG1, ARG2, etc variables. Sorry.
    In trunk (1.6 and higher), we have made all arguments local variables to
a macro call. They will not affect channel variables of the same name. This
includes the ARG1, ARG2, etc variables.
    Users can declare their own local variables by using the keyword 'local'
before setting them to a value;

                                             76


  Macro myroutine(firstarg, secondarg)
  {
    local Myvar=1;
    NoOp(Myvar is set to ${Myvar}, and firstarg is ${firstarg}, and secondarg is ${secondarg});
  }
   In the above example, Myvar, firstarg, and secondarg are all local vari-
ables, and will not be visible to the calling code, be it an extension, or another
Macro.
   If you need to make a local variable within the Set() application, you can
do it this way:
  Macro myroutine(firstarg, secondarg)
  {
    Set(LOCAL(Myvar)=1);
    NoOp(Myvar is set to ${Myvar}, and firstarg is ${firstarg}, and secondarg is ${secondarg});
  }

4.9.10 Loops
AEL has implementations of 'for' and 'while' loops.
  context loops {
      1 => {
            for (x=0; ${x} < 3; x=${x} + 1) {
                 Verbose(x is ${x} !);
            }
      }
      2 => {
            y=10;
            while (${y} >= 0) {
                 Verbose(y is ${y} !);
                 y=${y}-1;
            }
      }
  }

   NOTE: The conditional expression (the "${y} >= 0" above) is wrapped
in $[ ] so it can be evaluated. NOTE: The for loop test expression (the "$x
< 3" above) is wrapped in $[ ] so it can be evaluated.

4.9.11 Conditionals
AEL supports if and switch statements, like AEL, but adds ifTime, and ran-
dom. Unlike the original AEL, though, you do NOT need to put curly braces
around a single statement in the "true" branch of an if(), the random(), or
an ifTime() statement. The if(), ifTime(), and random() statements allow
optional else clause.

                                              77


 context conditional {
     _8XXX => {
          Dial(SIP/${EXTEN});
          if ("${DIALSTATUS}" = "BUSY")
          {
               NoOp(yessir);
               Voicemail(${EXTEN},b);
          }
          else
               Voicemail(${EXTEN},u);
          ifTime (14:00-25:00,sat-sun,*,*)
               Voicemail(${EXTEN},b);
          else
          {
               Voicemail(${EXTEN},u);
               NoOp(hi, there!);
          }
          random(51) NoOp(This should appear 51% of the time);
          random( 60 )
          {
                        NoOp( This should appear 60% of the time );
          }
          else
          {
                        random(75)
                        {
                                 NoOp( This should appear 30% of the time! );
                        }
                        else
                        {
                                 NoOp( This should appear 10% of the time! );
                        }
           }
     }
     _777X => {
          switch (${EXTEN}) {
               case 7771:
                    NoOp(You called 7771!);
                    break;
               case 7772:
                    NoOp(You called 7772!);
                    break;
               case 7773:
                    NoOp(You called 7773!);
                    // fall thru-
               pattern 777[4-9]:
                     NoOp(You called 777 something!);
               default:
                    NoOp(In the default clause!);
          }
     }
 }

   NOTE: The conditional expression in if() statements (the "${DIALSTATUS}"
= "BUSY" above) is wrapped by the compiler in $[] for evaluation.

                                           78


    NOTE: Neither the switch nor case values are wrapped in $[ ]; they can
be constants, or ${var} type references only.
    NOTE: AEL generates each case as a separate extension. case clauses
with no terminating 'break', or 'goto', have a goto inserted, to the next
clause, which creates a 'fall thru' effect.
    NOTE: AEL introduces the ifTime keyword/statement, which works just
like the if() statement, but the expression is a time value, exactly like that
used by the application GotoIfTime(). See Asterisk cmd GotoIfTime
    NOTE: The pattern statement makes sure the new extension that is cre-
ated has an ' ' preceding it to make sure asterisk recognizes the extension
name as a pattern.
    NOTE: Every character enclosed by the switch expression's parenthesis
are included verbatim in the labels generated. So watch out for spaces!
    NOTE: NEW: Previous to version 0.13, the random statement used the
"Random()" application, which has been deprecated. It now uses the RAND()
function instead, in the GotoIf application.

4.9.12 Break, Continue, and Return
Three keywords, break, continue, and return, are included in the syntax to
provide flow of control to loops, and switches.
    The break can be used in switches and loops, to jump to the end of the
loop or switch.
    The continue can be used in loops (while and for) to immediately jump
to the end of the loop. In the case of a for loop, the increment and test will
then be performed. In the case of the while loop, the continue will jump to
the test at the top of the loop.
    The return keyword will cause an immediate jump to the end of the
context, or macro, and can be used anywhere.

4.9.13 goto, jump, and labels
This is an example of how to do a goto in AEL.
  context gotoexample {
      s => {
  begin:
           NoOp(Infinite Loop! yay!);
           Wait(1);
           goto begin; // go to label in same extension

                                             79


       }
       3 => {
               goto s,begin; // go to label in different extension
        }
        4 => {
               goto gotoexample,s,begin; // overkill go to label in same context
        }
   }
   context gotoexample2 {
        s => {
      end:
              goto gotoexample,s,begin; // go to label in different context
        }
   }

    You can use the special label of "1" in the goto and jump statements.
It means the "first" statement in the extension. I would not advise trying
to use numeric labels other than "1" in goto's or jumps, nor would I advise
declaring a "1" label anywhere! As a matter of fact, it would be bad form to
declare a numeric label, and it might conflict with the priority numbers used
internally by asterisk.
    The syntax of the jump statement is: jump extension[,priority][@context]
If priority is absent, it defaults to "1". If context is not present, it is assumed
to be the same as that which contains the "jump".
   context gotoexample {
       s => {
   begin:
            NoOp(Infinite Loop! yay!);
            Wait(1);
            jump s; // go to first extension in same extension
       }
       3 => {
               jump s,begin; // go to label in different extension
        }
        4 => {
               jump s,begin@gotoexample; // overkill go to label in same context
        }
   }
   context gotoexample2 {
        s => {
      end:
              jump s@gotoexample; // go to label in different context
        }
   }

    NOTE: goto labels follow the same requirements as the Goto() applica-
tion, except the last value has to be a label. If the label does not exist, you

                                             80


will have run-time errors. If the label exists, but in a different extension, you
have to specify both the extension name and label in the goto, as in: goto
s,z; if the label is in a different context, you specify context,extension,label.
There is a note about using goto's in a switch statement below...
    NOTE AEL introduces the special label "1", which is the beginning con-
text number for most extensions.

4.9.14 Macros
A macro is defined in its own block like this. The arguments to the macro
are specified with the name of the macro. They are then referred to by that
same name. A catch block can be specified to catch special extensions.

  macro std-exten( ext , dev ) {
          Dial(${dev}/${ext},20);
          switch(${DIALSTATUS) {
          case BUSY:
                  Voicemail(${ext},b);
                  break;
          default:
                  Voicemail(${ext},u);
          }
          catch a {
                  VoiceMailMain(${ext});
                  return;
          }
  }
    A macro is then called by preceding the macro name with an amper-
sand. Empty arguments can be passed simply with nothing between com-
ments(0.11).

  context example {
       _5XXX => &std-exten(${EXTEN}, "IAX2");
       _6XXX => &std-exten(, "IAX2");
       _7XXX => &std-exten(${EXTEN},);
       _8XXX => &std-exten(,);
  }

4.10 Examples
  context demo {
       s => {
            Wait(1);
            Answer();
            TIMEOUT(digit)=5;

                                             81


          TIMEOUT(response)=10;
 restart:
          Background(demo-congrats);
 instructions:
          for (x=0; ${x} < 3; x=${x} + 1) {
               Background(demo-instruct);
               WaitExten();
          }
     }
     2 => {
          Background(demo-moreinfo);
          goto s,instructions;
     }
     3 => {
          LANGUAGE()=fr;
          goto s,restart;
     }
     500 => {
          Playback(demo-abouttotry);
          Dial(IAX2/guest@misery.digium.com);
          Playback(demo-nogo);
          goto s,instructions;
     }
     600 => {
          Playback(demo-echotest);
          Echo();
          Playback(demo-echodone);
          goto s,instructions;
     }
     # => {
 hangup:
          Playback(demo-thanks);
          Hangup();
     }
     t => goto #,hangup;
     i => Playback(invalid);
 }


4.11 Semantic Checks

AEL, after parsing, but before compiling, traverses the dialplan tree, and
makes several checks:
   * Macro calls to non-existent macros.

   * Macro calls to contexts.

   * Macro calls with argument count not matching the definition.
   * application call to macro. (missing the '&')

                                           82


   * application calls to "GotoIf", "GotoIfTime", "while", "endwhile", "Ran-
      dom", and "execIf", will generate a message to consider converting the
      call to AEL goto, while, etc. constructs.
   * goto a label in an empty extension.

   * goto a non-existent label, either a within-extension, within-context, or
      in a different context, or in any included contexts. Will even check
      "sister" context references.
   * All the checks done on the time values in the dial plan, are done on
      the time values in the ifTime() and includes times: o the time range
      has to have two times separated by a dash; o the times have to be in
      range of 0 to 24 hours. o The weekdays have to match the internal list,
      if they are provided; o the day of the month, if provided, must be in
      range of 1 to 31; o the month name or names have to match those in
      the internal list.
   * (0.5) If an expression is wrapped in $[ ... ], and the compiler will wrap
      it again, a warning is issued.
   * (0.5) If an expression had operators (you know, +,-,*,/,issued. Maybe
      someone forgot to wrap a variable name?

   * (0.12) check for duplicate context names.
   * (0.12) check for abstract contexts that are not included by any context.
   * (0.13) Issue a warning if a label is a numeric value.

   There are a subset of checks that have been removed until the proposed
AAL (Asterisk Argument Language) is developed and incorporated into As-
terisk. These checks will be:
   * (if the application argument analyzer is working: the presence of the
      'j' option is reported as error.
   * if options are specified, that are not available in an application.

   * if you specify too many arguments to an application.
   * a required argument is not present in an application call.

                                            83


 * Switch-case using "known" variables that applications set, that does
   not cover all the possible values. (a "default" case will solve this prob-
   lem. Each "unhandled" value is listed.
 * a Switch construct is used, which is uses a known variable, and the
   application that would set that variable is not called in the same ex-
   tension. This is a warning only...

 * Calls to applications not in the "applist" database (installed in /var/
   lib/asterisk/applist" on most systems).
 * In an assignment statement, if the assignment is to a function, the
   function name used is checked to see if it one of the currently known
   functions. A warning is issued if it is not.

4.12 Differences with the original version of AEL
1. The $[...] expressions have been enhanced to include the ==, ||, and
   && operators. These operators are exactly equivalent to the =, |, and
   & operators, respectively. Why? So the C, Java, C++ hackers feel at
   home here.

2. It is more free-form. The newline character means very little, and is
   pulled out of the white-space only for line numbers in error messages.
3. It generates more error messages -- by this I mean that any difference
   between the input and the grammar are reported, by file, line number,
   and column.
4. It checks the contents of $[ ] expressions (or what will end up being $[
   ] expressions!) for syntax errors. It also does matching paren/bracket
   counts.
5. It runs several semantic checks after the parsing is over, but before the
   compiling begins, see the list above.

6. It handles #include "filepath" directives. -- ALMOST anywhere, in
   fact. You could easily include a file in a context, in an extension, or at
   the root level. Files can be included in files that are included in files,
   down to 50 levels of hierarchy...
                                          84


 7. Local Goto's inside Switch statements automatically have the extension
    of the location of the switch statement appended to them.
 8. A pretty printer function is available within pbx ael.so.

 9. In the utils directory, two standalone programs are supplied for debug-
    ging AEL files. One is called "aelparse", and it reads in the /etc/
    asterisk/extensions.ael file, and shows the results of syntax and
    semantic checking on stdout, and also shows the results of compilation
    to stdout. The other is "aelparse1", which uses the original ael compiler
    to do the same work, reading in "/etc/asterisk/extensions.ael",
    using the original 'pbx ael.so' instead.
10. AEL supports the "jump" statement, and the "pattern" statement in
    switch constructs. Hopefully these will be documented in the AEL
    README.
11. Added the "return" keyword, which will jump to the end of an exten-
    sion/Macro.
12. Added the ifTime (<time range>|<days of week>|<days of month>|<months>
    ) [else ] construct, which executes much like an if () statement, but the
    decision is based on the current time, and the time spec provided in the
    ifTime. See the example above. (Note: all the other time-dependent
    Applications can be used via ifTime)

13. Added the optional time spec to the contexts in the includes construct.
    See examples above.
14. You don't have to wrap a single "true" statement in curly braces, as in
    the original AEL. An "else" is attached to the closest if. As usual, be
    careful about nested if statements! When in doubt, use curlies!
15. Added the syntax [regexten] [hint(channel)] to precede an extension
    declaration. See examples above, under "Extension". The regexten
    keyword will cause the priorities in the extension to begin with 2 instead
    of 1. The hint keyword will cause its arguments to be inserted in the
    extension under the hint priority. They are both optional, of course,
    but the order is fixed at the moment-- the regexten must come before
    the hint, if they are both present.

                                           85


 16. Empty case/default/pattern statements will "fall thru" as expected.
      (0.6)
 17. A trailing label in an extension, will automatically have a NoOp()
      added, to make sure the label exists in the extension on Asterisk. (0.6)
 18. (0.9) the semicolon is no longer required after a closing brace! (i.e. "];"
      ===> "}". You can have them there if you like, but they are not
      necessary. Someday they may be rejected as a syntax error, maybe.

 19. (0.9) the // comments are not recognized and removed in the spots
      where expressions are gathered, nor in application call arguments. You
      may have to move a comment if you get errors in existing files.
 20. (0.10) the random statement has been added. Syntax: random (
      <expr> ) <lucky-statement> [ else <unlucky-statement> ]. The prob-
      ability of the lucky-statement getting executed is <expr>, which should
      evaluate to an integer between 0 and 100. If the <lucky-statement>
      isn't so lucky this time around, then the <unlucky-statement> gets
      executed, if it is present.

4.13 Hints and Bugs
The safest way to check for a null strings is to say $[ "${x}" = "" ] The
old way would do as shell scripts often do, and append something on both
sides, like this: $[ ${x}foo = foo ]. The trouble with the old way, is that, if x
contains any spaces, then problems occur, usually syntax errors. It is better
practice and safer wrap all such tests with double quotes! Also, there are
now some functions that can be used in a variable reference, ISNULL(), and
LEN(), that can be used to test for an empty string: ${ISNULL(${x})} or
$[ ${LEN(${x})} = 0 ].
    Assignment vs. Set(). Keep in mind that setting a variable to value can
be done two different ways. If you choose say 'x=y;', keep in mind that AEL
will wrap the right-hand-side with $[]. So, when compiled into extension
language format, the end result will be 'Set(x=$[y])'. If you don't want this
effect, then say "Set(x=y);" instead.



                                              86


4.14 The Full Power of AEL

A newcomer to Asterisk will look at the above constructs and descriptions,
and ask, "Where's the string manipulation functions?", "Where's all the cool
operators that other languages have to offer?", etc.
   The answer is that the rich capabilities of Asterisk are made available
through AEL, via:
   * Applications: See Asterisk - documentation of application commands
   * Functions: Functions were implemented inside ${ .. } variable refer-
     ences, and supply many useful capabilities.

   * Expressions: An expression evaluation engine handles items wrapped
     inside $[...]. This includes some string manipulation facilities, arith-
     metic expressions, etc.
   * Application Gateway Interface: Asterisk can fork external processes
     that communicate via pipe. AGI applications can be written in any
     language. Very powerful applications can be added this way.
   * Variables: Channels of communication have variables associated with
     them, and asterisk provides some global variables. These can be ma-
     nipulated and/or consulted by the above mechanisms.













                                        87




Chapter 5

SLA: Shared Line Appearances


5.1 Introduction
The "SLA" functionality in Asterisk is intended to allow a setup that emu-
lates a simple key system. It uses the various abstraction layers already built
into Asterisk to emulate key system functionality across various devices, in-
cluding IP channels.

5.2 Configuration

5.2.1 Summary
An SLA system is built up of virtual trunks and stations mapped to real
Asterisk devices. The configuration for all of this is done in three different
files: extensions.conf, sla.conf, and the channel specific configuration file such
as sip.conf or dahdi.conf.

5.2.2 Dialplan
The SLA implementation can automatically generate the dialplan necessary
for basic operation if the "autocontext" option is set for trunks and stations in
sla.conf. However, for reference, here is an automatically generated dialplan
to help with custom building of the dialplan to include other features, such
as voicemail (5.3.2).

                                          88


    However, note that there is a little bit of additional configuration needed
if the trunk is an IP channel. This is discussed in the section on trunks
(5.2.3).
    There are extensions for incoming calls on a specific trunk, which execute
the SLATrunk application, as well as incoming calls from a station, which
execute SLAStation. Note that there are multiple extensions for incoming
calls from a station. This is because the SLA system has to know whether
the phone just went off hook, or if the user pressed a specific line button.
    Also note that there is a hint for every line on every station. This lets
the SLA system control each individual light on every phone to ensure that
it shows the correct state of the line. The phones must subscribe to the state
of each of their line appearances.
    Please refer to the examples section for full dialplan samples for SLA.

5.2.3 Trunks
An SLA trunk is a mapping between a virtual trunk and a real Asterisk
device. This device may be an analog FXO line, or something like a SIP
trunk. A trunk must be configured in two places. First, configure the device
itself in the channel specific configuration file such as dahdi.conf or sip.conf.
Once the trunk is configured, then map it to an SLA trunk in sla.conf.
   [line1]
   type=trunk
   device=DAHDI/1
    Be sure to configure the trunk's context to be the same one that is set
for the "autocontext" option in sla.conf if automatic dialplan configuration is
used. This would be done in the regular device entry in dahdi.conf, sip.conf,
etc. Note that the automatic dialplan generation creates the SLATrunk()
extension at extension 's'. This is perfect for DAHDI channels that are FXO
trunks, for example. However, it may not be good enough for an IP trunk,
since the call coming in over the trunk may specify an actual number.
    If the dialplan is being built manually, ensure that calls coming in on a
trunk execute the SLATrunk() application with an argument of the trunk
name, as shown in the dialplan example before.
    IP trunks can be used, but they require some additional configuration to
work.
    For this example, let's say we have a SIP trunk called "mytrunk" that is
going to be used as line4. Furthermore, when calls come in on this trunk,

                                              89


they are going to say that they are calling the number "12564286000". Also,
let's say that the numbers that are valid for calling out this trunk are NANP
numbers, of the form 1NXXNXXXXXX.
    In sip.conf, there would be an entry for [mytrunk]. For [mytrunk], set
context=line4.
  [line4]
  type=trunk
  device=Local/disa@line4_outbound

  [line4]
  exten => 12564286000,1,SLATrunk(line4)
  [line4_outbound]
  exten => disa,1,Disa(no-password,line4_outbound)
  exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@mytrunk)
    So, when a station picks up their phone and connects to line 4, they are
connected to the local dialplan. The Disa application plays dialtone to the
phone and collects digits until it matches an extension. In this case, once
the phone dials a number like 12565551212, the call will proceed out the SIP
trunk.

5.2.4 Stations
An SLA station is a mapping between a virtual station and a real Asterisk
device. Currently, the only channel driver that has all of the features nec-
essary to support an SLA environment is chan sip. So, to configure a SIP
phone to use as a station, you must configure sla.conf and sip.conf.

  [station1]
  type=station
  device=SIP/station1
  trunk=line1
  trunk=line2
    Here are some hints on configuring a SIP phone for use with SLA:

   1. Add the SIP channel as a [station] in sla.conf.
   2. Configure the phone in sip.conf. If automatic dialplan configuration
       was used by enabling the "autocontext" option in sla.conf, then this
       entry in sip.conf should have the same context setting.


                                            90


  3. On the phone itself, there are various things that must be configured
      to make everything work correctly:
      Let's say this phone is called "station1" in sla.conf, and it uses trunks
      named "line1" and line2".
       (a) Two line buttons must be configured to subscribe to the state of
           the following extensions: - station1 line1 - station1 line2

       (b) The line appearance buttons should be configured to dial the ex-
           tensions that they are subscribed to when they are pressed.
       (c) If you would like the phone to automatically connect to a trunk
           when it is taken off hook, then the phone should be automatically
           configured to dial "station1" when it is taken off hook.

5.3 Configuration Examples

5.3.1 Basic SLA
This is an example of the most basic SLA setup. It uses the automatic
dialplan generation so the configuration is minimal.
   sla.conf:
  [line1]
  type=trunk
  device=DAHDI/1
  autocontext=line1
  [line2]
  type=trunk
  device=DAHDI/2
  autocontext=line2
  [station](!)
  type=station
  trunk=line1
  trunk=line2
  autocontext=sla_stations
  [station1](station)
  device=SIP/station1
  [station2](station)
  device=SIP/station2
  [station3](station)
  device=SIP/station3


                                           91


   With this configuration, the dialplan is generated automatically. The first
DAHDI channel should have its context set to "line1" and the second should
be set to "line2" in dahdi.conf. In sip.conf, station1, station2, and station3
should all have their context set to "sla stations".
   For reference, here is the automatically generated dialplan for this situa-
tion:
  [line1]
  exten => s,1,SLATrunk(line1)
  [line2]
  exten => s,2,SLATrunk(line2)

  [sla_stations]
  exten => station1,1,SLAStation(station1)
  exten => station1_line1,hint,SLA:station1_line1
  exten => station1_line1,1,SLAStation(station1_line1)
  exten => station1_line2,hint,SLA:station1_line2
  exten => station1_line2,1,SLAStation(station1_line2)
  exten => station2,1,SLAStation(station2)
  exten => station2_line1,hint,SLA:station2_line1
  exten => station2_line1,1,SLAStation(station2_line1)
  exten => station2_line2,hint,SLA:station2_line2
  exten => station2_line2,1,SLAStation(station2_line2)
  exten => station3,1,SLAStation(station3)
  exten => station3_line1,hint,SLA:station3_line1
  exten => station3_line1,1,SLAStation(station3_line1)
  exten => station3_line2,hint,SLA:station3_line2
  exten => station3_line2,1,SLAStation(station3_line2)

5.3.2 SLA and Voicemail

This is an example of how you could set up a single voicemail box for the
phone system. The voicemail box number used in this example is 1234, which
would be configured in voicemail.conf.
   For this example, assume that there are 2 trunks and 3 stations. The
trunks are DAHDI/1 and DAHDI/2. The stations are SIP/station1, SIP/station2,
and SIP/station3.
   In dahdi.conf, channel 1 has context=line1 and channel 2 has context=line2.
   In sip.conf, all three stations are configured with context=sla stations.
   When the stations pick up their phones to dial, they are allowed to dial
NANP numbers for outbound calls, or 8500 for checking voicemail.
   sla.conf:
  [line1]

                                            92


type=trunk
device=Local/disa@line1_outbound
[line2]
type=trunk
device=Local/disa@line2_outbound

[station](!)
type=station
trunk=line1
trunk=line2
[station1](station)
device=SIP/station1

[station2](station)
device=SIP/station2
[station3](station)
device=SIP/station3


 extensions.conf:

[macro-slaline]
exten => s,1,SLATrunk(${ARG1})
exten => s,n,Goto(s-${SLATRUNK_STATUS},1)
exten => s-FAILURE,1,Voicemail(1234,u)
exten => s-UNANSWERED,1,Voicemail(1234,u)
[line1]
exten => s,1,Macro(slaline,line1)

[line2]
exten => s,2,Macro(slaline,line2)
[line1_outbound]
exten => disa,1,Disa(no-password,line1_outbound)
exten => _1NXXNXXXXXX,1,Dial(DAHDI/1/${EXTEN})
exten => 8500,1,VoicemailMain(1234)

[line2_outbound]
exten => disa,1,Disa(no-password|line2_outbound)
exten => _1NXXNXXXXXX,1,Dial(DAHDI/2/${EXTEN})
exten => 8500,1,VoicemailMain(1234)
[sla_stations]

exten => station1,1,SLAStation(station1)
exten => station1_line1,hint,SLA:station1_line1
exten => station1_line1,1,SLAStation(station1_line1)
exten => station1_line2,hint,SLA:station1_line2
exten => station1_line2,1,SLAStation(station1_line2)
exten => station2,1,SLAStation(station2)
exten => station2_line1,hint,SLA:station2_line1
exten => station2_line1,1,SLAStation(station2_line1)


                                         93


  exten => station2_line2,hint,SLA:station2_line2
  exten => station2_line2,1,SLAStation(station2_line2)
  exten => station3,1,SLAStation(station3)
  exten => station3_line1,hint,SLA:station3_line1
  exten => station3_line1,1,SLAStation(station3_line1)
  exten => station3_line2,hint,SLA:station3_line2
  exten => station3_line2,1,SLAStation(station3_line2)


5.4 Call Handling
5.4.1 Summary
This section is intended to describe how Asterisk handles calls inside of the
SLA system so that it is clear what behavior is expected.

5.4.2 Station goes off hook (not ringing)

When a station goes off hook, it should initiate a call to Asterisk with the
extension that indicates that the phone went off hook without specifying
a specific line. In the examples in this document, for the station named
"station1", this extension is simply named, "station1".
    Asterisk will attempt to connect this station to the first available trunk
that is not in use. Asterisk will check the trunks in the order that they were
specified in the station entry in sla.conf. If all trunks are in use, the call will
be denied.
    If Asterisk is able to acquire an idle trunk for this station, then trunk is
connected to the station and the station will hear dialtone. The station can
then proceed to dial a number to call. As soon as a trunk is acquired, all
appearances of this line on stations will show that the line is in use.

5.4.3 Station goes off hook (ringing)
When a station goes off hook while it is ringing, it should simply answer
the call that had been initiated to it to make it ring. Once the station has
answered, Asterisk will figure out which trunk to connect it to. It will connect
it to the highest priority trunk that is currently ringing. Trunk priority is
determined by the order that the trunks are listed in the station entry in
sla.conf.

                                             94


5.4.4 Line button on a station is pressed
When a line button is pressed on a station, the station should initiate a call
to Asterisk with the extension that indicates which line button was pressed.
In the examples given in this document, for a station named "station1" and
a trunk named "line1", the extension would be "station1 line1".
    If the specified trunk is not in use, then the station will be connected to
it and will hear dialtone. All appearances of this trunk will then show that
it is now in use.
    If the specified trunk is on hold by this station, then this station will be
reconnected to the trunk. The line appearance for this trunk on this station
will now show in use. If this was the only station that had the call on hold,
then all appearances of this trunk will now show that it is in use. Otherwise,
all stations that are not currently connected to this trunk will show it on
hold.
    If the specified trunk is on hold by a different station, then this station
will be connected to the trunk only if the trunk itself and the station(s)
that have it on hold do not have private hold enabled. If connected, the
appeareance of this trunk on this station will then show in use. All stations
that are not currently connected to this trunk will show it on hold.















                                            95




Chapter 6

Channel Drivers


6.1 IAX2

6.1.1 Introduction
This section is intended as an introduction to the Inter-Asterisk eXchange
v2 (or simply IAX2) protocol. It provides both a theoretical background and
practical information on its use.

6.1.2 Why IAX2?
The first question most people are thinking at this point is "Why do you
need another VoIP protocol? Why didn't you just use SIP or H.323?"
   Well, the answer is a fairly complicated one, but in a nutshell it's like
this... Asterisk is intended as a very flexible and powerful communications
tool. As such, the primary feature we need from a VoIP protocol is the ability
to meet our own goals with Asterisk, and one with enough flexibility that
we could use it as a kind of laboratory for inventing and implementing new
concepts in the field. Neither H.323 or SIP fit the roles we needed, so we
developed our own protocol, which, while not standards based, provides a
number of advantages over both SIP and H.323, some of which are:
   * Interoperability with NAT/PAT/Masquerade firewalls
         -- IAX seamlessly interoperates through all sorts of NAT and PAT
            and other firewalls, including the ability to place and receive calls,

                                          96


       and transfer calls to other stations.
* High performance, low overhead protocol

    -- When running on low-bandwidth connections, or when running
       large numbers of calls, optimized bandwidth utilization is imper-
       ative. IAX uses only 4 bytes of overhead
* Internationalization support
    -- IAX transmits language information, so that remote PBX content
       can be delivered in the native language of the calling party.
* Remote dialplan polling

    -- IAX allows a PBX or IP phone to poll the availability of a number
       from a remote server. This allows PBX dialplans to be centralized.
* Flexible authentication
    -- IAX supports cleartext, md5, and RSA authentication, providing
       flexible security models for outgoing calls and registration services.

* Multimedia protocol
    -- IAXsupports the transmission of voice, video, images, text, HTML,
       DTMF, and URL's. Voice menus can be presented in both audibly
       and visually.
* Call statistic gathering

    -- IAX gathers statistics about network performance (including la-
       tency and jitter, as well as providing end-to-end latency measure-
       ment.
* Call parameter communication
    -- Caller*ID, requested extension, requested context, etc are all com-
       municated through the call.

* Single socket design

                                      97


         -- IAX's single socket design allows up to 32768 calls to be multi-
             plexed.
   While we value the importance of standards based (i.e. SIP) call handling,
hopefully this will provide a reasonable explanation of why we developed IAX
rather than starting with SIP.

6.1.3 Configuration
For examples of a configuration, please see the iax.conf.sample in your the
/configs directory of you source code distribution.

6.1.4 IAX2 Jitterbuffer
The new jitterbuffer
You must add "jitterbuffer=yes" to either the [general] part of iax.conf, or
to a peer or a user. (just like the old jitterbuffer). Also, you can set "maxjit-
terbuffer=n", which puts a hard-limit on the size of the jitterbuffer of "n
milliseconds". It is not necessary to have the new jitterbuffer on both sides
of a call; it works on the receive side only.

PLC
The new jitterbuffer detects packet loss. PLC is done to try to recreate these
lost packets in the codec decoding stage, as the encoded audio is translated
to slinear. PLC is also used to mask jitterbuffer growth.
   This facility is enabled by default in iLBC and speex, as it has no addi-
tional cost. This facility can be enabled in adpcm, alaw, g726, gsm, lpc10,
and ulaw by setting genericplc => true in the [plc] section of codecs.conf.

Trunktimestamps
To use this, both sides must be using Asterisk v1.2 or later. Setting "trunk-
timestamps=yes" in iax.conf will cause your box to send 16-bit timestamps
for each trunked frame inside of a trunk frame. This will enable you to use
jitterbuffer for an IAX2 trunk, something that was not possible in the old
architecture.


                                            98


   The other side must also support this functionality, or else, well, bad
things will happen. If you don't use trunktimestamps, there's lots of ways
the jitterbuffer can get confused because timestamps aren't necessarily sent
through the trunk correctly.

Communication with Asterisk v1.0.x systems
You can set up communication with v1.0.x systems with the new jitterbuffer,
but you can't use trunks with trunktimestamps in this communication.
   If you are connecting to an Asterisk server with earlier versions of the
software (1.0.x), do not enable both jitterbuffer and trunking for the involved
peers/users in order to be able to communicate. Earlier systems will not
support trunktimestamps.
   You may also compile chan iax2.c without the new jitterbuffer, enabling
the old backwards compatible architecture. Look in the source code for
instructions.
Testing and monitoring

You can test the effectiveness of PLC and the new jitterbuffer's detection
of loss by using the new CLI command "iax2 test losspct <n>". This will
simulate n percent packet loss coming in to chan iax2. You should find
that with PLC and the new JB, 10 percent packet loss should lead to just a
tiny amount of distortion, while without PLC, it would lead to silent gaps in
your audio.
   "iax2 show netstats" shows you statistics for each iax2 call you have up.
The columns are "RTT" which is the round-trip time for the last PING, and
then a bunch of s tats for both the local side (what you're receiving), and the
remote side (what the other end is telling us they are seeing). The remote
stats may not be complete if the remote end isn't using the new jitterbuffer.
   The stats shown are:
   * Jit: The jitter we have measured (milliseconds)
   * Del: The maximum delay imposed by the jitterbuffer (milliseconds)
   * Lost: The number of packets we've detected as lost.

   * %: The percentage of packets we've detected as lost recently.

                                            99


   * Drop: The number of packets we've purposely dropped (to lower la-
     tency).
   * OOO: The number of packets we've received out-of-order

   * Kpkts: The number of packets we've received / 1000.
Reporting problems
There's a couple of things that can make calls sound bad using the jitter-
buffer:

  1. The JB and PLC can make your calls sound better, but they can't fix
     everything. If you lost 10 frames in a row, it can't possibly fix that. It
     really can't help much more than one or two consecutive frames.
  2. Bad timestamps: If whatever is generating timestamps to be sent to
     you generates nonsensical timestamps, it can confuse the jitterbuffer.
     In particular, discontinuities in timestamps will really upset it: Things
     like timestamps sequences which go 0, 20, 40, 60, 80, 34000, 34020,
     34040, 34060... It's going to think you've got about 34 seconds of jitter
     in this case, etc.. The right solution to this is to find out what's causing
     the sender to send us such nonsense, and fix that. But we should also
     figure out how to make the receiver more robust in cases like this.
     chan iax2 will actually help fix this a bit if it's more than 3 seconds or
     so, but at some point we should try to think of a better way to detect
     this kind of thing and resynchronize.
     Different clock rates are handled very gracefully though; it will actually
     deal with a sender sending 20% faster or slower than you expect just
     fine.

  3. Really strange network delays: If your network "pauses" for like 5
     seconds, and then when it restarts, you are sent some packets that
     are 5 seconds old, we are going to see that as a lot of jitter. We
     already throw away up to the worst 20 frames like this, though, and
     the "maxjitterbuffer" parameter should put a limit on what we do in
     this case.


                                          100


6.2 mISDN

6.2.1 Introduction
This package contains the mISDN Channel Driver for the Asterisk PBX. It
supports every mISDN Hardware and provides an interface for Asterisk.
6.2.2 Features
   * NT and TE mode

   * PP and PMP mode
   * BRI and PRI (with BNE1 and BN2E1 Cards)
   * Hardware bridging
   * DTMF detection in HW+mISDNdsp
   * Display messages on phones (on those that support it)
   * app SendText

   * HOLD/RETRIEVE/TRANSFER on ISDN phones : )
   * Allow/restrict user number presentation
   * Volume control
   * Crypting with mISDNdsp (Blowfish)
   * Data (HDLC) callthrough

   * Data calling (with app ptyfork +pppd)
   * Echo cancellation
   * Call deflection
   * Some others



                                     101


6.2.3 Fast Installation Guide
It is easy to install mISDN and mISDNuser. This can be done by:

    * You can download latest stable releases from http://www.misdn.org/downloads/
    * Just fetch the newest head of the GIT (mISDN project moved from
       CVS)In details this process described here: http://www.misdn.org/index.php/GIT
    then compile and install both with:
  cd mISDN ;
  make && make install
(you will need at least your kernel headers to compile mISDN).
  cd mISDNuser ;
  make && make install
Now you can compile chan misdn, just by making Asterisk:
  cd asterisk ;
  ./configure && make && make install

That's all!
    Follow the instructions in the mISDN Package for how to load the Kernel
Modules. Also install process described in http://www.misdn.org/index.php/Installing_mISDN

6.2.4 Pre-Requisites
To compile and install this driver, you'll need at least one mISDN Driver and
the mISDNuser package. Chan misdn works with both, the current release
version and the development (svn trunk) version of Asterisk.
    You should use Kernels >= 2.6.9

6.2.5 Configuration
First of all you must configure the mISDN drivers, please follow the instruc-
tions in the mISDN package to do that, the main config file and config script
is:
  /etc/init.d/misdn-init and
  /etc/misdn-init.conf
Now you will want to configure the misdn.conf file which resides in the As-
terisk config directory (normally /etc/asterisk).

                                           102


misdn.conf: [general]
The misdn.conf file contains a "general" subsection, and user subsections
which contain misdn port settings and different Asterisk contexts.
    In the general subsection you can set options that are not directly port
related. There is for example the very important debug variable which you
can set from the Asterisk cli (command line interface) or in this configuration
file, bigger numbers will lead to more debug output. There's also a trace file
option, which takes a path+filename where debug output is written to.

misdn.conf: [default] subsection
The default subsection is another special subsection which can contain all
the options available in the user/port subsections. The user/port subsections
inherit their parameters from the default subsection.

misdn.conf: user/port subsections
The user subsections have names which are unequal to "general". Those
subsections contain the ports variable which mean the mISDN Ports. Here
you can add multiple ports, comma separated.
    Especially for TE-Mode Ports there is a msns option. This option tells
the chan misdn driver to listen for incoming calls with the given msns, you
can insert a '*' as single msn, which leads to getting every incoming call.
If you want to share on PMP TE S0 with Asterisk and a phone or ISDN
card you should insert here the msns which you assign to Asterisk. Finally a
context variable resides in the user subsections, which tells chan misdn where
to send incoming calls to in the Asterisk dial plan (extension.conf).

Dial and Options String
The dial string of chan misdn got more complex, because we added more
features, so the generic dial string looks like:
   mISDN/<port>[:bchannel]|g:<group>/<extension>[/<OPTIONSSTRING>]
   The Optionsstring looks Like:
   :<optchar><optarg>:<optchar><optarg>...
   the ":" character is the delimiter.
   The available options are:

                                            103


    a - Have Asterisk detect DTMF tones on called channel
    c - Make crypted outgoing call, optarg is keyindex
    d - Send display text to called phone, text is the optarg
    e - Perform echo cancelation on this channel,
        takes taps as optarg (32,64,128,256)
   e! - Disable echo cancelation on this channel
    f - Enable fax detection
    h - Make digital outgoing call
   h1 - Make HDLC mode digital outgoing call
    i - Ignore detected DTMF tones, don't signal them to Asterisk,
        they will be transported inband.
   jb - Set jitter buffer length, optarg is length
   jt - Set jitter buffer upper threshold, optarg is threshold
   jn - Disable jitter buffer
    n - Disable mISDN DSP on channel.
        Disables: echo cancel, DTMF detection, and volume control.
    p - Caller ID presentation,
        optarg is either 'allowed' or 'restricted'
    s - Send Non-inband DTMF as inband
   vr - Rx gain control, optarg is gain
   vt - Tx gain control, optarg is gain

   chan misdn registers a new dial plan application "misdn set opt" when
loaded. This application takes the Optionsstring as argument. The Syntax
is:
misdn_set_opt(<OPTIONSSTRING>)

   When you set options in the dialstring, the options are set in the external
channel. When you set options with misdn set opt, they are set in the current
incoming channel. So if you like to use static encryption, the scenario looks
as follows:

Phone1 --> * Box 1 --> PSTN_TE
PSTN_TE --> * Box 2 --> Phone2

   The encryption must be done on the PSTN sides, so the dialplan on the
boxes are:

* Box 1:
exten => _${CRYPT_PREFIX}X.,1,Dial(mISDN/g:outbound/:c1)
* Box 2:
exten => ${CRYPT_MSN},1,misdn_set_opt(:c1)
exten => ${CRYPT_MSN},2,dial(${PHONE2})


                                           104


6.2.6 mISDN CLI commands

At the Asterisk cli you can try to type in:
misdn <tab> <tab>

    Now you should see the misdn cli commands:
  - clean
    -> pid (cleans a broken call, use with care, leads often
         to a segmentation fault)
  - send
    -> display (sends a Text Message to a Asterisk channel,
         this channel must be an misdn channel)
  - set
    -> debug (sets debug level)
  - show
    -> config (shows the configuration options)
    -> channels (shows the current active misdn channels)
    -> channel (shows details about the given misdn channels)
    -> stacks (shows the current ports, their protocols and states)
    -> fullstacks (shows the current active and inactive misdn channels)
  - restart
    -> port (restarts given port (L2 Restart) )

  - reload (reloads misdn.conf)
    You can only use "misdn send display" when an Asterisk channel is cre-
ated and isdn is in the correct state. "correct state" means that you have
established a call to another phone (must not be isdn though).
    Then you use it like this:
    misdn send display mISDN/1/101 "Hello World!"
    where 1 is the Port of the Card where the phone is plugged in, and 101
is the msn (callerid) of the Phone to send the text to.

6.2.7 mISDN Variables

mISDN Exports/Imports a few Variables:

- MISDN_ADDRESS_COMPLETE : Is either set to 1 from the Provider, or you
           can set it to 1 to force a sending complete.




                                            105


6.2.8 Debugging and sending bug reports
If you encounter problems, you should set up the debugging flag, usually
debug=2 should be enough. The messages are divided into Asterisk and
mISDN parts. mISDN Debug messages begin with an 'I', Asterisk messages
begin with an '*', the rest is clear I think.
    Please take a trace of the problem and open a report in the Asterisk issue
tracker at http://bugs.digium.com in the "channel drivers" project, "chan -
misdn" category. Read the bug guidelines to make sure you provide all the
information needed.

6.2.9 Examples

Here are some examples of how to use chan misdn in the dialplan (exten-
sions.conf):
  [globals]
  OUT_PORT=1 ; The physical Port of the Card
  OUT_GROUP=ExternE1 ; The Group of Ports defined in misdn.conf
  [misdnIn]
  exten => _X.,1,Dial(mISDN/${OUT_PORT}/${EXTEN})
  exten => _0X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1})
  exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello)
  exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello Test:n)
    On the last line, you will notice the last argument (Hello); this is sent as
Display Message to the Phone.

6.2.10 Known Problems

Q: I cannot hear any tone after a successful CONNECT to the other end.
    A: You forgot to load mISDNdsp, which is now needed by chan misdn for
switching and DTMF tone detection.

6.3 Local
6.3.1 Introduction

In Asterisk, Local channels are a method used to treat an extension in the
dialplan as if it were an external device. In essense, Asterisk will send the

                                            106


call back into the dialplan as the destination of the call, versus sending the
call to a device.
    Two of the most common areas where Local channels are used include
members configured for queues, and in use with callfiles. There are also other
uses where you want to ring two destinations, but with different information,
such as different callerID for each outgoing request.

6.3.2 Examples
Local channels are best demonstrated through the use of an example. Our
first example isn't terribly useful, but will demonstrate how Local channels
can execute dialplan logic by dialing from the Dial() application.


6.3.3 Trivial Local channel example
In our dialplan (extensions.conf), we can Dial() another part of the dialplan
through the use Local channels. To do this, we can use the following dialplan:
  [devices]
  exten => 201,1,Verbose(2,Dial another part of the dialplan via the Local chan)
  exten => 201,n,Verbose(2,Outside channel: ${CHANNEL})
  exten => 201,n,Dial(Local/201@extensions)
  exten => 201,n,Hangup()
  [extensions]
  exten => 201,1,Verbose(2,Made it to the Local channel)
  exten => 201,n,Verbose(2,Inside channel: ${CHANNEL})
  exten => 201,n,Dial(SIP/some-named-extension,30)
  exten => 201,n,Hangup()

    The output of the dialplan would look something like the following. The
output has been broken up with some commentary to explain what we're
looking at.
  -- Executing [201@devices:1] Verbose("SIP/my_desk_phone-00000014", "2,Dial another part of the dialplan via the
               Local chan") in new stack
  == Dial another part of the dialplan via the Local chan

    We dial extension 201 from SIP/my desk phone which has entered the
[devices] context. The first line simply outputs some information via the
Verbose() application.
  -- Executing [201@devices:2] Verbose("SIP/my_desk_phone-00000014",
                    "2,Outside channel: SIP/my_desk_phone-00000014") in new stack
  == Outside channel: SIP/my_desk_phone-00000014

                                            107


    The next line is another Verbose() application statement that tells us our
current channel name. We can see that the channel executing the current
dialplan is a desk phone (aptly named 'my desk phone').
  -- Executing [201@devices:3] Dial("SIP/my_desk_phone-00000014", "Local/201@extensions") in new stack
  -- Called 201@extensions

    Now the third step in our dialplan executes the Dial() application which
calls extension 201 in the [extensions] context of our dialplan. There is no
requirement that we use the same extension number -- we could have just
as easily used a named extension, or some other number. Remember that
we're dialing another channel, but instead of dialing a device, we're "dialing"
another part of the dialplan.
  -- Executing [201@extensions:1] Verbose("Local/201@extensions-7cf4;2", "2,Made it to the Local
                channel") in new stack
  == Made it to the Local channel

    Now we've verified we've dialed another part of the dialplan. We can see
the channel executing the dialplan has changed to Local/201@extensions-
7cf4;2. The part '-7cf4;2' is just the unique identifier, and will be different
for you.
  -- Executing [201@extensions:2] Verbose("Local/201@extensions-7cf4;2", "2,Inside channel:
                Local/201@extensions-7cf4;2") in new stack
  == Inside channel: Local/201@extensions-7cf4;2

    Here we use the Verbose() application to see what our current channel
name is. As you can see the current channel is a Local channel which we
created from our SIP channel.
  -- Executing [201@extensions:3] Dial("Local/201@extensions-7cf4;2", "SIP/some-named-extension,30") in new stack

    And from here, we're using another Dial() application to call a SIP device
configured in sip.conf as [some-named-extension].
    Now that we understand a simple example of calling the Local channel,
let's expand upon this example by using Local channels to call two devices
at the same time, but delay calling one of the devices.




                                            108


6.3.4 Delay dialing devices
Lets say when someone calls extension 201, we want to ring both the desk
phone and their cellphone at the same time, but we want to wait about 6
seconds to start dialing the cellphone. This is useful in a situation when
someone might be sitting at their desk, but don't want both devices ringing
at the same time, but also doesn't want to wait for the full ring cycle to
execute on their desk phone before rolling over to their cellphone.
   The dialplan for this would look something like the following:

  [devices]
  exten => 201,1,Verbose(2,Call desk phone and cellphone but with delay)
  exten => 201,n,Dial(Local/deskphone-201@extensions&Local/cellphone-201@extensions,30)
  exten => 201,n,Voicemail(201@default,${IF($[${DIALSTATUS} = BUSY]?b:u)})
  exten => 201,n,Hangup()
  [extensions]
  ; Dial the desk phone
  exten => deskphone-201,1,Verbose(2,Dialing desk phone of extension 201)
  exten => deskphone-201,n,Dial(SIP/0004f2040001) ; SIP device with MAC address
                                                   ; of 0004f2040001
  ; Dial the cellphone
  exten => cellphone-201,1,Verbose(2,Dialing cellphone of extension 201)
  exten => cellphone-201,n,Verbose(2,-- Waiting 6 seconds before dialing)
  exten => cellphone-201,n,Wait(6)
  exten => cellphone-201,n,Dial(DAHDI/g0/14165551212)

   When someone dials extension 201 in the [devices] context, it will execute
the Dial() application, and call two Local channels at the same time:
   * Local/deskphone-201@extensions

   * Local/cellphone-201@extensions
   It will then ring both of those extensions for 30 seconds before rolling over
to the Voicemail() application and playing the appropriate voicemail record-
ing depending on whether the ${DIALSTATUS} variable returned BUSY or
not.
   When reaching the deskphone-201 extension, we execute the Dial() ap-
plication which calls the SIP device configured as '0004f204001' (the MAC
address of the device). When reaching the cellphone-201 extension, we dial
the cellphone via the DAHDI channel using group zero (g0) and dialing phone
number 1-416-555-1212.


                                           109


6.3.5 Dialing destinations with different information
With Asterisk, we can place a call to multiple destinations by separating
the technology/destination pair with an ampersand (&). For example, the
following Dial() line would ring two separate destinations for 30 seconds:

  exten => 201,1,Dial(SIP/0004f2040001&DAHDI/g0/14165551212,30)
   That line would dial both the SIP/0004f2040001 device (likely a SIP
device on the network) and dial the phone number 1-416-555-1212 via a
DAHDI interface. In our example though, we would be sending the same
callerID information to both end points, but perhaps we want to send a
different callerID to one of the destinations?
   We can send different callerIDs to each of the destinations if we want by
using the Local channel. The following example shows how this is possible
because we would Dial() two different Local channels from our top level
Dial(), and that would then execute some dialplan before sending the call off
to the final destinations.
  [devices]
  exten => 201,1,NoOp()
  exten => 201,n,Dial(Local/201@internal&Local/201@external,30)
  exten => 201,n,Voicemail(201@default,${IF($[${DIALSTATUS} = BUSY]?b:u)})
  exten => 201,n,Hangup()

  [internal]
  exten => 201,1,Verbose(2,Placing internal call for extension 201)
  exten => 201,n,Set(CALLERID(name)=From Sales)
  exten => 201,n,Dial(SIP/0004f2040001,30)
  [external]
  exten => 201,1,Verbose(2,Placing external call for extension 201)
  exten => 201,n,Set(CALLERID(name)=Acme Cleaning)
  exten => 201,n,Dial(DAHDI/g0/14165551212)
   With the dialplan above, we've sent two different callerIDs to the desti-
nations:

   * "From Sales" was sent to the local device SIP/0004f2040001
   * "Acme Cleaning" was sent to the remote number 1-416-555-1212 via
      DAHDI

   Because each of the channels is independent from the other, you could
perform any other call manipulation you need. Perhaps the 1-416-555-1212
                                           110


number is a cell phone and you know you can only ring that device for 18
seconds before the voicemail would pick up. You could then limit the length
of time the external number is dialed, but still allow the internal device to
be dialed for a longer period of time.

6.3.6 Using callfiles and Local channels
Another example is to use callfiles and Local channels so that you can execute
some dialplan prior to performing a Dial(). We'll construct a callfile which
will then utilize a Local channel to lookup a bit of information in the AstDB
and then place a call via the channel configured in the AstDB.
    First, lets construct our callfile that will use the Local channel to do some
lookups prior to placing our call. More information on constructing callfiles
is located in the doc/callfiles.txt file of your Asterisk source.
    Our callfile will simply look like the following:
Channel: Local/201@devices
Application: Playback
Data: silence/1&tt-weasels

    Add the callfile information to a file such as 'callfile.new' or some other
appropriately named file.
    Our dialplan will perform a lookup in the AstDB to determine which
device to call, and will then call the device, and upon answer, Playback() the
silence/1 (1 second of silence) and the tt-weasels sound files.
    Before looking at our dialplan, lets put some data into AstDB that we
can then lookup from the dialplan. From the Asterisk CLI, run the following
command:
*CLI> database put phones 201/device SIP/0004f2040001
    We've now put the device destination (SIP/0004f2040001) into the 201/de-
vice key within the phones family. This will allow us to lookup the device
location for extension 201 from the database.
    We can then verify our entry in the database using the 'database show'
CLI command:

*CLI> database show
/phones/201/device : SIP/0004f2040001
                                            111


   Now lets create the dialplan that will allow us to call SIP/0004f2040001
when we request extension 201 from the [extensions] context via our Local
channel.
  [devices]
  exten => 201,1,NoOp()
  exten => 201,n,Set(DEVICE=${DB(phones/${EXTEN}/device)})
  exten => 201,n,GotoIf($[${ISNULL(${DEVICE})}]?hangup) ; if nothing returned,
                                                        ; then hangup
  exten => 201,n,Dial(${DEVICE},30)
  exten => 201,n(hangup(),Hangup()

   Then, we can perform a call to our device using the callfile by moving it
into the /var/spool/asterisk/outgoing/ directory.

# mv callfile.new /var/spool/asterisks/outgoing
   Then after a moment, you should see output on your console similar to
the following, and your device ringing. Information about what is going on
during the output has also been added throughout.
      -- Attempting call on Local/201@devices for application Playback(silence/1&tt-weasels) (Retry 1)

   You'll see the line above as soon as Asterisk gets the request from the
callfile.
  -- Executing [201@devices:1] NoOp("Local/201@devices-ecf0;2", "") in new stack
  -- Executing [201@devices:2] Set("Local/201@devices-ecf0;2", "DEVICE=SIP/0004f2040001") in new stack

   This is where we performed our lookup in the AstDB. The value of
SIP/0004f2040001 was then returned and saved to the DEVICE channel vari-
able.
      -- Executing [201@devices:3] GotoIf("Local/201@devices-ecf0;2", "0?hangup") in new stack

   We perform a check to make sure ${DEVICE} isn't NULL. If it is, we'll
just hangup here.
      -- Executing [201@devices:4] Dial("Local/201@devices-ecf0;2", "SIP/0004f2040001,30") in new stack
      -- Called 000f2040001
      -- SIP/0004f2040001-00000022 is ringing

   Now we call our device SIP/0004f2040001 from the Local channel.
     -- SIP/0004f2040001-00000022 answered Local/201@devices-ecf0;2

                                           112


   We answer the call.
         > Channel Local/201@devices-ecf0;1 was answered.
         > Launching Playback(silence/1&tt-weasels) on Local/201@devices-ecf0;1

   We then start playing back the files.
      -- <Local/201@devices-ecf0;1> Playing 'silence/1.slin' (language 'en')
    == Spawn extension (devices, 201, 4) exited non-zero on 'Local/201@devices-ecf0;2'

   At this point we now see the Local channel has been optimized out of
the call path. This is important as we'll see in examples later. By default,
the Local channel will try to optimize itself out of the call path as soon as it
can. Now that the call has been established and audio is flowing, it gets out
of the way.
      -- <SIP/0004f2040001-00000022> Playing 'tt-weasels.ulaw' (language 'en')
  [Mar 1 13:35:23] NOTICE[16814]: pbx_spool.c:349 attempt_thread: Call completed to Local/201@devices

   We can now see the tt-weasels file is played directly to the destination
(instead of through the Local channel which was optimized out of the call
path) and then a NOTICE stating the call was completed.

6.3.7 Understanding When To Use /n
Lets take a look at an example that demonstrates when the use of the /n
directive is necessary. If we spawn a Local channel which does a Dial() to a
SIP channel, but we use the L() option (which is used to limit the amount of
time a call can be active, along with warning tones when the time is nearly
up), it will be associated with the Local channel, which is then optimized
out of the call path, and thus won't perform as expected.
   This following dialplan will not perform as expected.
[services]
exten => 2,1,Dial(SIP/PHONE\_B,,L(60000:45000:15000))

[internal]
exten => 4,1,Dial(Local/2@services)



                                            113


   By default, the Local channel will try to optimize itself out of the call
path. This means that once the Local channel has established the call be-
tween the destination and Asterisk, the Local channel will get out of the way
and let Asterisk and the end point talk directly, instead of flowing through
the Local channel.
   This can have some adverse effects when you're expecting information
to be available during the call that gets associated with the Local channel.
When the Local channel is optimized out of the call path, any Dial() flags, or
channel variables associated with the Local channel are also destroyed and
are no longer available to Asterisk.
   We can force the Local channel to remain in the call path by utilizing the
/n directive. By adding /n to the end of the channel definition, we can keep
the Local channel in the call path, along with any channel variables, or other
channel specific information.
   In order to make this behave as we expect (limiting the call), we would
change:
[internal]
exten => 4,1,Dial(Local/2@services)

   ...into the following:
[internal]
exten => 4,1,Dial(Local/2@services/n)
   By adding /n to the end, our Local channel will now stay in the call path
and not go away.
   Why does adding the /n option all of a suddon make the 'L' option work?
First we need to show an overview of the call flow that doesn't work properly,
and discuss the information associated with the channels:

  1. SIP device PHONE A calls Asterisk via a SIP INVITE
  2. Asterisk accepts the INVITE and then starts processing dialplan logic
      in the [internal] context
  3. Our dialplan calls Dial(Local/2@services) -- notice no /n
  4. The Local channel then executes dialplan at extension 2 within the
      [services] context

                                          114


  5. Extension 2 within [services] then performs Dial() to PHONE B with
     the line: Dial(SIP/PHONE B,,L(60000:45000:15000))
  6. SIP/PHONE B then answers the call
  7. Even though the L option was given when dialing the SIP device, the
     L information is stored in the channel that is doing the Dial() which is
     the Local channel, and not the endpoint SIP channel.

  8. The Local channel in the middle, containing the information for track-
     ing the time allowance of the call, is then optimized out of the call
     path, losing all information about when to terminate the call.
  9. SIP/PHONE A and SIP/PHONE B then continue talking indefinitely.
   Now, if we were to add /n to our dialplan at step three (3) then we
would force the Local channel to stay in the call path, and the L() option
associated with the Dial() from the Local channel would remain, and our
warning sounds and timing would work as expected.
   There are two workarounds for the above described scenario:

  1. Use what we just described, Dial(Local/2@services/n) to cause the Lo-
     cal channel to remain in the call path so that the L() option used inside
     the Local channel is not discarded when optimization is performed.
  2. Place the L() option at the outermost part of the path so that when
     the middle is optimized out of the call path, the information required
     to make L() work is associated with the outside channel. The L infor-
     mation will then be stored on the calling channel, which is PHONE A.
     For example:

     [services]
     exten => 2,1,Dial(SIP/PHONE_B)
     [internal]
     exten => 4,1,Dial(Local/2@services,,L(60000:45000:15000));



                                          115


6.3.8 Local channel modifiers
There are additional modifiers for the Local channel as well. They include:

   * 'n' -- Adding "/n" at the end of the string will make the Local channel
     not do a native transfer (the "n" stands for "n"o release) upon the
     remote end answering the line. This is an esoteric, but important
     feature if you expect the Local channel to handle calls exactly like a
     normal channel. If you do not have the "no release" feature set, then
     as soon as the destination (inside of the Local channel) answers the
     line and one audio frame passes, the variables and dial plan will revert
     back to that of the original call, and the Local channel will become a
     zombie and be removed from the active channels list. This is desirable
     in some circumstances, but can result in unexpected dialplan behavior
     if you are doing fancy things with variables in your call handling.
   * 'j' -- Adding "/j" at the end of the string allows you to use the generic
     jitterbuffer on incoming calls going to Asterisk applications. For exam-
     ple, this would allow you to use a jitterbuffer for an incoming SIP call
     to Voicemail by putting a Local channel in the middle. The 'j' option
     must be used in conjunction with the 'n' option to make sure that the
     Local channel does not get optimized out of the call.
     This option is available starting in the Asterisk 1.6.0 branch.
   * 'm' -- Using the "/m" option will cause the Local channel to forward
     music on hold (MoH) start and stop requests. Normally the Local
     channel acts on them and it is started or stopped on the Local channel
     itself. This options allows those requests to be forwarded through the
     Local channel.
     This option is available starting in the Asterisk 1.4 branch.

   * 'b' -- The "/b" option causes the Local channel to return the actual
     channel that is behind it when queried. This is useful for transfer sce-
     narios as the actual channel will be transferred, not the Local channel.
     This option is available starting in the Asterisk 1.6.0 branch.



                                            116



Chapter 7


Distributed Universal Number
Discovery (DUNDi)



7.1 Introduction
http://www.dundi.com
   Mark Spencer, Digium, Inc.
   DUNDi is essentially a trusted, peer-to-peer system for being able to call
any phone number from the Internet. DUNDi works by creating a network of
nodes called the "DUNDi E.164 Trust Group" which are bound by a common
peering agreement known as the General Peering Agreement or GPA. The
GPA legally binds the members of the Trust Group to provide good-faith
accurate information to the other nodes on the network, and provides stan-
dards by which the community can insure the integrity of the information on
the nodes themselves. Unlike ENUM or similar systems, DUNDi is explicitly
designed to preclude any necessity for a single centralized system which could
be a source of fees, regulation, etc.
   Much less dramatically, DUNDi can also be used within a private enter-
prise to share a dialplan efficiently between multiple nodes, without incurring
a risk of a single point of failure. In this way, administrators can locally add
extensions which become immediately available to the other nodes in the
system.
   For more information visit http://www.dundi.com


                                          117


7.2 DUNDIQUERY and DUNDIRESULT

The DUNDIQUERY and DUNDIRESULT dialplan functions will let you
initiate a DUNDi query from the dialplan, see how many results there are,
and access each one. Here is some example usage:
  exten => 1,1,Set(ID=${DUNDIQUERY(1,dundi_test,b)})
  exten => 1,n,Set(NUM=${DUNDIRESULT(${ID},getnum)})
  exten => 1,n,NoOp(There are ${NUM} results)
  exten => 1,n,Set(X=1)
  exten => 1,n,While($[${X} <= ${NUM}])
  exten => 1,n,NoOp(Result ${X} is ${DUNDIRESULT(${ID},${X})})
  exten => 1,n,Set(X=$[${X} + 1])
  exten => 1,n,EndWhile

7.3 Peering Agreement

                            DIGIUM GENERAL PEERING AGREEMENT (TM)
                               Version 1.0.0, September 2004
 Copyright (C) 2004 Digium, Inc.
                       445 Jan Davis Drive, Huntsville, AL 35806 USA

 Everyone is permitted to copy and distribute complete verbatim copies
 of this General Peering Agreement provided it is not modified in any
 manner.
           ------------------------------------------------------

                            DIGIUM GENERAL PEERING AGREEMENT

                                              PREAMBLE
  For most of the history of telecommunications, the power of being able
to locate and communicate with another person in a system, be it across
a hall or around the world, has always centered around a centralized
authority -- from a local PBX administrator to regional and national
RBOCs, generally requiring fees, taxes or regulation. By contrast,
DUNDi is a technology developed to provide users the freedom to
communicate with each other without the necessity of any centralized

                                           118


authority. This General Peering Agreement ("GPA") is used by individual
parties (each, a "Participant") to allow them to build the E164 trust
group for the DUNDi protocol.

  To protect the usefulness of the E164 trust group for those who use
it, while keeping the system wholly decentralized, it is necessary to
replace many of the responsibilities generally afforded to a company or
government agency, with a set of responsibilities implemented by the
parties who use the system, themselves. It is the goal of this document
to provide all the protections necessary to keep the DUNDi E164 trust
group useful and reliable.
  The Participants wish to protect competition, promote innovation and
value added services and make this service valuable both commercially
and non-commercially. To that end, this GPA provides special terms and
conditions outlining some permissible and non-permissible revenue
sources.

  This GPA is independent of any software license or other license
agreement for a program or technology employing the DUNDi protocol. For
example, the implementation of DUNDi used by Asterisk is covered under a
separate license. Each Participant is responsible for compliance with
any licenses or other agreements governing use of such program or
technology that they use to peer.
  You do not have to execute this GPA to use a program or technology
employing the DUNDi protocol, however if you do not execute this GPA,
you will not be able to peer using DUNDi and the E164 context with
anyone who is a member of the trust group by virtue of their having
executed this GPA with another member.

The parties to this GPA agree as follows:

  0. DEFINITIONS. As used herein, certain terms shall be defined as
follows:
    (a) The term "DUNDi" means the DUNDi protocol as published by
        Digium, Inc. or its successor in interest with respect to the

                              119


    DUNDi protocol specification.
(b) The terms "E.164" and "E164" mean ITU-T specification E.164 as
    published by the International Telecommunications Union (ITU) in
    May, 1997.

(c) The term "Service" refers to any communication facility (e.g.,
    telephone, fax, modem, etc.), identified by an E.164-compatible
    number, and assigned by the appropriate authority in that
    jurisdiction.

(d) The term "Egress Gateway" refers an Internet facility that
    provides a communications path to a Service or Services that may
    not be directly addressable via the Internet.
(e) The term "Route" refers to an Internet address, policies, and
    other characteristics defined by the DUNDi protocol and
    associated with the Service, or the Egress Gateway which
    provides access to the specified Service.

(f) The term "Propagate" means to accept or transmit Service and/or
    Egress Gateway Routes only using the DUNDi protocol and the
    DUNDi context "e164" without regard to case, and does not apply
    to the exchange of information using any other protocol or
    context.
(g) The term "Peering System" means the network of systems that
    Propagate Routes.

(h) The term "Subscriber" means the owner of, or someone who
    contracts to receive, the services identified by an E.164
    number.

(i) The term "Authorizing Individual" means the Subscriber to a
    number who has authorized a Participant to provide Routes
    regarding their services via this Peering System.
(j) The term "Route Authority" refers to a Participant that provides

                          120


        an original source of said Route within the Peering System.
        Routes are propagated from the Route Authorities through the
        Peering System and may be cached at intermediate points. There
        may be multiple Route Authorities for any Service.

    (k) The term "Participant" (introduced above) refers to any member
        of the Peering System.
    (l) The term "Service Provider" refers to the carrier (e.g.,
        exchange carrier, Internet Telephony Service Provider, or other
        reseller) that provides communication facilities for a
        particular Service to a Subscriber, Customer or other End User.

    (m) The term "Weight" refers to a numeric quality assigned to a
        Route as per the DUNDi protocol specification. The current
        Weight definitions are shown in Exhibit A.
  1. PEERING. The undersigned Participants agree to Propagate Routes
with each other and any other member of the Peering System and further
agree not to Propagate DUNDi Routes with a third party unless they have
first have executed this GPA (in its unmodified form) with such third
party. The Participants further agree only to Propagate Routes with
Participants whom they reasonably believe to be honoring the terms of
the GPA. Participants may not insert, remove, amend, or otherwise
modify any of the terms of the GPA.

  2. ACCEPTABLE USE POLICY. The DUNDi protocol contains information
that reflect a Subscriber's or Egress Gateway's decisions to receive
calls. In addition to the terms and conditions set forth in this GPA,
the Participants agree to honor the intent of restrictions encoded in
the DUNDi protocol. To that end, Participants agree to the following:
    (a) A Participant may not utilize or permit the utilization of
        Routes for which the Subscriber or Egress Gateway provider has
        indicated that they do not wish to receive "Unsolicited Calls"
        for the purpose of making an unsolicited phone call on behalf of
        any party or organization.


                              121


    (b) A Participant may not utilize or permit the utilization of
        Routes which have indicated that they do not wish to receive
        "Unsolicited Commercial Calls" for the purpose of making an
        unsolicited phone call on behalf of a commercial organization.

    (c) A Participant may never utilize or permit the utilization of any
        DUNDi route for the purpose of making harassing phone calls.
    (d) A Party may not utilize or permit the utilization of DUNDi
        provided Routes for any systematic or random calling of numbers
        (e.g., for the purpose of locating facsimile, modem services, or
        systematic telemarketing).

    (e) Initial control signaling for all communication sessions that
        utilize Routes obtained from the Peering System must be sent
        from a member of the Peering System to the Service or Egress
        Gateway identified in the selected Route. For example, 'SIP
        INVITES' and IAX2 "NEW" commands must be sent from the
        requesting DUNDi node to the terminating Service.
    (f) A Participant may not disclose any specific Route, Service or
        Participant contact information obtained from the Peering System
        to any party outside of the Peering System except as a
        by-product of facilitating communication in accordance with
        section 2e (e.g., phone books or other databases may not be
        published, but the Internet addresses of the Egress Gateway or
        Service does not need to be obfuscated.)

    (g) The DUNDi Protocol requires that each Participant include valid
        contact information about itself (including information about
        nodes connected to each Participant). Participants may use or
        disclose the contact information only to ensure enforcement of
        legal furtherance of this Agreement.

  3. ROUTES. The Participants shall only propagate valid Routes, as
defined herein, through the Peering System, regardless of the original
source. The Participants may only provide Routes as set forth below,
and then only if such Participant has no good faith reason to believe
                              122


such Route to be invalid or unauthorized.
    (a) A Participant may provide Routes if each Route has as its
        original source another member of the Peering System who has
        duly executed the GPA and such Routes are provided in accordance
        with this Agreement; provided that the Routes are not modified
        (e.g., with regards to existence, destination, technology or
        Weight); or

    (b) A Participant may provide Routes for Services with any Weight
        for which it is the Subscriber; or

    (c) A Participant may provide Routes for those Services whose
        Subscriber has authorized the Participant to do so, provided
        that the Participant is able to confirm that the Authorizing
        Individual is the Subscriber through:
            i. a written statement of ownership from the Authorizing
               Individual, which the Participant believes in good faith
               to be accurate (e.g., a phone bill with the name of the
               Authorizing Individual and the number in question); or

           ii. the Participant's own direct personal knowledge that the
               Authorizing Individual is the Subscriber.
    (d) A Participant may provide Routes for Services, with Weight in
        accordance with the Current DUNDi Specification, if it can in
        good faith provide an Egress Gateway to that Service on the
        traditional telephone network without cost to the calling party.

  4. REVOCATION. A Participant must provide a free, easily accessible
mechanism by which a Subscriber may revoke permission to act as a Route
Authority for his Service. A Participant must stop acting as a Route
Authority for that Service within 7 days after:

    (a) receipt of a revocation request;
    (b) receiving other notice that the Service is no longer valid; or

                              123


    (c) determination that the Subscriber's information is no longer
        accurate (including that the Subscriber is no longer the service
        owner or the service owner's authorized delegate).

  5. SERVICE FEES. A Participant may charge a fee to act as a Route
Authority for a Service, with any Weight, provided that no Participant
may charge a fee to propagate the Route received through the Peering
System.
  6. TOLL SERVICES. No Participant may provide Routes for any Services
that require payment from the calling party or their customer for
communication with the Service. Nothing in this section shall prohibit
a Participant from providing routes for Services where the calling party
may later enter into a financial transaction with the called party
(e.g., a Participant may provide Routes for calling cards services).

  7. QUALITY. A Participant may not intentionally impair communication
using a Route provided to the Peering System (e.g. by adding delay,
advertisements, reduced quality). If for any reason a Participant is
unable to deliver a call via a Route provided to the Peering System,
that Participant shall return out-of-band Network Congestion
notification (e.g. "503 Service Unavailable" with SIP protocol or
"CONGESTION" with IAX protocol).
  8. PROTOCOL COMPLIANCE. Participants agree to Propagate Routes in
strict compliance with current DUNDi protocol specifications.

  9. ADMINISTRATIVE FEES. A Participant may charge (but is not required
to charge) another Participant a reasonable fee to cover administrative
expenses incurred in the execution of this Agreement. A Participant may
not charge any fee to continue the relationship or to provide Routes to
another Participant in the Peering System.
  10. CALLER IDENTIFICATION. A Participant will make a good faith effort
to ensure the accuracy and appropriate nature of any caller
identification that it transmits via any Route obtained from the Peering
System. Caller identification shall at least be provided as a valid

                              124


E.164 number.
  11. COMPLIANCE WITH LAWS. The Participants are solely responsible for
determining to what extent, if any, the obligations set forth in this
GPA conflict with any laws or regulations their region. A Participant
may not provide any service or otherwise use DUNDi under this GPA if
doing so is prohibited by law or regulation, or if any law or regulation
imposes requirements on the Participant that are inconsistent with the
terms of this GPA or the Acceptable Use Policy.

  12. WARRANTY. EACH PARTICIPANT WARRANTS TO THE OTHER PARTICIPANTS THAT
IT MADE, AND WILL CONTINUE TO MAKE, A GOOD FAITH EFFORT TO AUTHENTICATE
OTHERS IN THE PEERING SYSTEM AND TO PROVIDE ACCURATE INFORMATION IN
ACCORDANCE WITH THE TERMS OF THIS GPA. THIS WARRANTY IS MADE BETWEEN
THE PARTICIPANTS, AND THE PARTICIPANTS MAY NOT EXTEND THIS WARRANTY TO
ANY NON-PARTICIPANT INCLUDING END-USERS.
  13. DISCLAIMER OF WARRANTIES. THE PARTICIPANTS UNDERSTAND AND AGREE
THAT ANY SERVICE PROVIDED AS A RESULT OF THIS GPA IS "AS IS." EXCEPT FOR
THOSE WARRANTIES OTHERWISE EXPRESSLY SET FORTH HEREIN, THE PARTICIPANTS
DISCLAIM ANY REPRESENTATIONS OR WARRANTIES OF ANY KIND OR NATURE,
EXPRESS OR IMPLIED, AS TO THE CONDITION, VALUE OR QUALITIES OF THE
SERVICES PROVIDED HEREUNDER, AND SPECIFICALLY DISCLAIM ANY
REPRESENTATION OR WARRANTY OF MERCHANTABILITY, SUITABILITY OR FITNESS
FOR A PARTICULAR PURPOSE OR AS TO THE CONDITION OR WORKMANSHIP THEREOF,
OR THE ABSENCE OF ANY DEFECTS THEREIN, WHETHER LATENT OR PATENT,
INCLUDING ANY WARRANTIES ARISING FROM A COURSE OF DEALING, USAGE OR
TRADE PRACTICE. EXCEPT AS EXPRESSLY PROVIDED HEREIN, THE PARTICIPANTS
EXPRESSLY DISCLAIM ANY REPRESENTATIONS OR WARRANTIES THAT THE PEERING
SERVICE WILL BE CONTINUOUS, UNINTERRUPTED OR ERROR-FREE, THAT ANY DATA
SHARED OR OTHERWISE MADE AVAILABLE WILL BE ACCURATE OR COMPLETE OR
OTHERWISE COMPLETELY SECURE FROM UNAUTHORIZED ACCESS.
  14. LIMITATION OF LIABILITIES. NO PARTICIPANT SHALL BE LIABLE TO ANY
OTHER PARTICIPANT FOR INCIDENTAL, INDIRECT, CONSEQUENTIAL, SPECIAL,
PUNITIVE OR EXEMPLARY DAMAGES OF ANY KIND (INCLUDING LOST REVENUES OR
PROFITS, LOSS OF BUSINESS OR LOSS OF DATA) IN ANY WAY RELATED TO THIS
GPA, WHETHER IN CONTRACT OR IN TORT, REGARDLESS OF WHETHER SUCH

                              125


PARTICIPANT WAS ADVISED OF THE POSSIBILITY THEREOF.
  15. END-USER AGREEMENTS. The Participants may independently enter
into agreements with end-users to provide certain services (e.g., fees
to a Subscriber to originate Routes for that Service). To the extent
that provision of these services employs the Peering System, the Parties
will include in their agreements with their end-users terms and
conditions consistent with the terms of this GPA with respect to the
exclusion of warranties, limitation of liability and Acceptable Use
Policy. In no event may a Participant extend the warranty described in
Section 12 in this GPA to any end-users.

  16. INDEMNIFICATION. Each Participant agrees to defend, indemnify and
hold harmless the other Participant or third-party beneficiaries to this
GPA (including their affiliates, successors, assigns, agents and
representatives and their respective officers, directors and employees)
from and against any and all actions, suits, proceedings,
investigations, demands, claims, judgments, liabilities, obligations,
liens, losses, damages, expenses (including, without limitation,
attorneys' fees) and any other fees arising out of or relating to (i)
personal injury or property damage caused by that Participant, its
employees, agents, servants, or other representatives; (ii) any act or
omission by the Participant, its employees, agents, servants or other
representatives, including, but not limited to, unauthorized
representations or warranties made by the Participant; or (iii) any
breach by the Participant of any of the terms or conditions of this GPA.

  17. THIRD PARTY BENEFICIARIES. This GPA is intended to benefit those
Participants who have executed the GPA and who are in the Peering
System. It is the intent of the Parties to this GPA to give to those
Participants who are in the Peering System standing to bring any
necessary legal action to enforce the terms of this GPA.
  18. TERMINATION. Any Participant may terminate this GPA at any time,
with or without cause. A Participant that terminates must immediately
cease to Propagate.

  19. CHOICE OF LAW. This GPA and the rights and duties of the Parties

                              126


hereto shall be construed and determined in accordance with the internal
laws of the State of New York, United States of America, without regard
to its conflict of laws principles and without application of the United
Nations Convention on Contracts for the International Sale of Goods.
  20. DISPUTE RESOLUTION. Unless otherwise agreed in writing, the
exclusive procedure for handling disputes shall be as set forth herein.
Notwithstanding such procedures, any Participant may, at any time, seek
injunctive relief in addition to the process described below.

    (a) Prior to mediation or arbitration the disputing Participants
        shall seek informal resolution of disputes. The process shall be
        initiated with written notice of one Participant to the other
        describing the dispute with reasonable particularity followed
        with a written response within ten (10) days of receipt of
        notice. Each Participant shall promptly designate an executive
        with requisite authority to resolve the dispute. The informal
        procedure shall commence within ten (10) days of the date of
        response. All reasonable requests for non-privileged information
        reasonably related to the dispute shall be honored. If the
        dispute is not resolved within thirty (30) days of commencement
        of the procedure either Participant may proceed to mediation or
        arbitration pursuant to the rules set forth in (b) or (c) below.

    (b) If the dispute has not been resolved pursuant to (a) above or,
        if the disputing Participants fail to commence informal dispute
        resolution pursuant to (a) above, either Participant may, in
        writing and within twenty (20) days of the response date noted
        in (a) above, ask the other Participant to participate in a one
        (1) day mediation with an impartial mediator, and the other
        Participant shall do so. Each Participant will bear its own
        expenses and an equal share of the fees of the mediator. If the
        mediation is not successful the Participants may proceed with
        arbitration pursuant to (c) below.
    (c) If the dispute has not been resolved pursuant to (a) or (b)
        above, the dispute shall be promptly referred, no later than one
        (1) year from the date of original notice and subject to

                              127


applicable statute of limitations, to binding arbitration in
accordance with the UNCITRAL Arbitration Rules in effect on the
date of this contract. The appointing authority shall be the
International Centre for Dispute Resolution. The case shall be
administered by the International Centre for Dispute Resolution
under its Procedures for Cases under the UNCITRAL Arbitration
Rules. Each Participant shall bear its own expenses and shall
share equally in fees of the arbitrator. All arbitrators shall
have substantial experience in information technology and/or in
the telecommunications business and shall be selected by the
disputing participants in accordance with UNCITRAL Arbitration
Rules. If any arbitrator, once selected is unable or unwilling
to continue for any reason, replacement shall be filled via the
process described above and a re-hearing shall be conducted. The
disputing Participants will provide each other with all
requested documents and records reasonably related to the
dispute in a manner that will minimize the expense and
inconvenience of both parties. Discovery will not include
depositions or interrogatories except as the arbitrators
expressly allow upon a showing of need. If disputes arise
concerning discovery requests, the arbitrators shall have sole
and complete discretion to resolve the disputes. The parties and
arbitrator shall be guided in resolving discovery disputes by
the Federal Rules of Civil Procedure. The Participants agree
that time of the essence principles shall guide the hearing and
that the arbitrator shall have the right and authority to issue
monetary sanctions in the event of unreasonable delay. The
arbitrator shall deliver a written opinion setting forth
findings of fact and the rationale for the award within thirty
(30) days following conclusion of the hearing. The award of the
arbitrator, which may include legal and equitable relief, but
which may not include punitive damages, will be final and
binding upon the disputing Participants, and judgment may be
entered upon it in accordance with applicable law in any court
having jurisdiction thereof. In addition to award the
arbitrator shall have the discretion to award the prevailing
Participant all or part of its attorneys' fees and costs,
including fees associated with arbitrator, if the arbitrator

                      128


        determines that the positions taken by the other Participant on
        material issues of the dispute were without substantial
        foundation. Any conflict between the UNCITRAL Arbitration Rules
        and the provisions of this GPA shall be controlled by this GPA.

  21. INTEGRATED AGREEMENT. This GPA, constitutes the complete
integrated agreement between the parties concerning the subject matter
hereof. All prior and contemporaneous agreements, understandings,
negotiations or representations, whether oral or in writing, relating to
the subject matter of this GPA are superseded and canceled in their
entirety.
  22. WAIVER. No waiver of any of the provisions of this GPA shall be
deemed or shall constitute a waiver of any other provision of this GPA,
whether or not similar, nor shall such waiver constitute a continuing
waiver unless otherwise expressly so provided in writing. The failure
of either party to enforce at any time any of the provisions of this
GPA, or the failure to require at any time performance by either party
of any of the provisions of this GPA, shall in no way be construed to be
a present or future waiver of such provisions, nor in any way affect the
ability of a Participant to enforce each and every such provision
thereafter.

  23. INDEPENDENT CONTRACTORS. Nothing in this GPA shall make the
Parties partners, joint venturers, or otherwise associated in or with
the business of the other. Parties are, and shall always remain,
independent contractors. No Participant shall be liable for any debts,
accounts, obligations, or other liabilities of the other Participant,
its agents or employees. No party is authorized to incur debts or other
obligations of any kind on the part of or as agent for the other. This
GPA is not a franchise agreement and does not create a franchise
relationship between the parties, and if any provision of this GPA is
deemed to create a franchise between the parties, then this GPA shall
automatically terminate.
  24. CAPTIONS AND HEADINGS. The captions and headings used in this GPA
are used for convenience only and are not to be given any legal effect.


                              129


  25. EXECUTION. This GPA may be executed in counterparts, each of which
so executed will be deemed to be an original and such counterparts
together will constitute one and the same Agreement. The Parties shall
transmit to each other a signed copy of the GPA by any means that
faithfully reproduces the GPA along with the Signature. For purposes of
this GPA, the term "signature" shall include digital signatures as
defined by the jurisdiction of the Participant signing the GPA.
                         Exhibit A

Weight Range Requirements

0-99 May only be used under authorization of Owner
100-199 May only be used by the Owner's service
                        provider, regardless of authorization.

200-299 Reserved -- do not use for e164 context.
300-399 May only be used by the owner of the code under
                        which the Owner's number is a part of.

400-499 May be used by any entity providing access via
                        direct connectivity to the Public Switched
                        Telephone Network.

500-599 May be used by any entity providing access via
                        indirect connectivity to the Public Switched
                        Telephone Network (e.g. Via another VoIP
                        provider)
600- Reserved-- do not use for e164 context.

                 Participant Participant
Company:

Address:
                              130


Email:

          _________________________ _________________________
          Authorized Signature Authorized Signature

Name:


END OF GENERAL PEERING AGREEMENT
------------------------------------------------

How to Peer using this GPA If you wish to exchange routing information
with parties using the e164 DUNDi context, all you must do is execute
this GPA with any member of the Peering System and you will become a
member of the Peering System and be able to make Routes available in
accordance with this GPA.

DUNDi, IAX, Asterisk and GPA are trademarks of Digium, Inc.















                              131




Chapter 8

ENUM


8.1 The ENUMLOOKUP dialplan function
The ENUMLOOKUP function is more complex than it first may appear,
and this guide is to give a general overview and set of examples that may
be well-suited for the advanced user to evaluate in their consideration of
ENUM or ENUM-like lookup strategies. This document assumes a famil-
iarity with ENUM (RFC3761) or ENUM-like methods, as well as familiarity
with NAPTR DNS records (RFC2915, RFC3401-3404). For an overview
of NAPTR records, and the use of NAPTRs in the ENUM global phone-
number-to-DNSmapping scheme, please see http://www.voip-info.org/tiki-index.php?page=ENUM
for more detail.
   Using ENUM within Asterisk can be simple or complex, depending on
how many failover methods and redundancy procedures you wish to utilize.
Implementation of ENUM paths is supposedly defined by the person creat-
ing the NAPTR records, but the local administrator may choose to ignore
certain NAPTR response methods (URI types) or prefer some over others,
which is in contradiction to the RFC. The ENUMLOOKUP method simply
provides administrators a method for determining NAPTR results in either
the globally unique ENUM (e164.arpa) DNS tree, or in other ENUM-like
DNS trees which are not globally unique. The methods to actually create
channels ("dial") results given by the ENUMLOOKUP function is then up to
the administrator to implement in a way that best suits their environment.

Function: ENUMLOOKUP(number[,Method-type[,options[,record#[,zone-suffix]]]])
                                         132


   Performs an ENUM tree lookup on the specified number, method type,
and ordinal record offset, and returns one of four different values:
  1. post-parsed NAPTR of one method (URI) type

  2. count of elements of one method (URI) type
  3. count of all method types
  4. full URI of method at a particular point in the list of all possible meth-
     ods

8.1.1 Arguments
   * number
        -- telephone number or search string. Only numeric values within
           this string are parsed; all other digits are ignored for search, but
           are re-written during NAPTR regexp expansion.

   * service type
        -- tel, sip, h323, iax2, mailto, ...[any other string], ALL. Default type
           is "sip". Special name of "ALL" will create a list of method types
           across all NAPTR records for the search number, and then put the
           results in an ordinal list starting with 1. The position <number>
           specified will then be returned, starting with 1 as the first record
           (lowest value) in the list. The service types are not hardcoded
           in Asterisk except for the default (sip) if no other service type
           specified; any method type string (IANA-approved or not) may
           be used except for the string "ALL".
   * options
        -- c
              * count. Returns the number of records of this type are returned
                 (regardless of order or priority.) If "ALL" is the specified
                 service type, then a count of all methods will be returned for
                 the DNS record.

   * record#
                                           133


          -- which record to present if multiple answers are returned <integer>
             = The record in priority/order sequence based on the total count
             of records passed back by the query. If a service type is specified,
             all entries of that type will be sorted into an ordinal list starting
             with 1 (by order first, then priority). The default of <options> is
             "1"
    * zone suffix

          -- allows customization of the ENUM zone. Default is e164.arpa.
8.1.2 Examples

Let's use this ENUM list as an example (note that these examples exist in
the DNS, and will hopefully remain in place as example destinations, but
they may change or become invalid over time. The end result URIs are not
guaranteed to actually work, since some of these hostnames or SIP proxies
are imaginary. Of course, the tel: replies go to directory assistance for New
York City and San Francisco...) Also note that the complex SIP NAPTR at
weight 30 will strip off the leading "+" from the dialed string if it exists. This
is probably a better NAPTR than hard-coding the number into the NAPTR,
and it is included as a more complex regexp example, though other simpler
NAPTRs will work just as well.
0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 10 100 "u"
       "E2U+tel" "!^\\+13015611020$!tel:+12125551212!" .
0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 21 100 "u"
       "E2U+tel" "!^\\+13015611020$!tel:+14155551212!" .
0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 25 100 "u"
       "E2U+sip" "!^\\+13015611020$!sip:2203@sip.fox-den.com!" .
0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 26 100 "u"
       "E2U+sip" "!^\\+13015611020$!sip:1234@sip-2.fox-den.com!" .
0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 30 100 "u"
       "E2U+sip" "!^\\+*([^\\*]*)!sip:\\1@sip-3.fox-den.com!" .
0.2.0.1.1.6.5.1.0.3.1.loligo.com. 3600 IN NAPTR 55 100 "u"
       "E2U+mailto" "!^\\+13015611020$!mailto:jtodd@fox-den.com!" .
    Example 1: Simplest case, using first SIP return (use all defaults except
for domain name)

                                             134


exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,,,,loligo.com)})
  returns: ${foo}="2203@sip.fox-den.com"
    Example 2: What is the first "tel" pointer type for this number? (after
sorting by order/preference; default of "1" is assumed in options field)
exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,tel,,,loligo.com)})
  returns: ${foo}="+12125551212"

    Example 3: How many "sip" pointer type entries are there for this num-
ber?
exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,sip,c,,loligo.com)})
  returns: ${foo}=3
    Example 4: For all the "tel" pointer type entries, what is the second one
in the list? (after sorting by preference)
exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,tel,,2,loligo.com)})
  returns: ${foo}="+14155551212"

    Example 5: How many NAPTRs (tel, sip, mailto, etc.) are in the list for
this number?
exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,ALL,c,,loligo.com)})
  returns: ${foo}=6
    Example 6: Give back the second full URI in the sorted list of all NAPTR
URIs:
exten => 100,1,Set(foo=${ENUMLOOKUP(+13015611020,ALL,,2,loligo.com)})
  returns: ${foo}="tel:+14155551212" [note the "tel:" prefix in the string]

    Example 7: Look up first SIP entry for the number in the e164.arpa zone
(all defaults)
exten => 100,1,Set(foo=${ENUMLOOKUP(+437203001721)})
  returns: ${foo}="enum-test@sip.nemox.net" [note: this result is
  subject to change as it is "live" DNS and not under my control]
    Example 8: Look up the ISN mapping in freenum.org alpha test zone


                                          135


exten => 100,1,Set(foo=${ENUMLOOKUP(1234*256,,,,freenum.org)})
  returns: ${foo}="1234@204.91.156.10" [note: this result is subject
  to change as it is "live" DNS]
  Example 9: Give back the first SIP pointer for a number in the

enum.yoydynelabs.com zone (invalid lookup)
exten => 100,1,Set(foo=${ENUMLOOKUP(1234567890,sip,,1,enum.yoyodynelabs.com)})
  returns: ${foo}=""
8.1.3 Usage notes and subtle features
  * The use of "+" in lookups is confusing, and warrants further explana-
    tion. All E.164 numbers ("global phone numbers") by definition need
    a leading "+" during ENUM lookup. If you neglect to add a leading
    "+", you may discover that numbers that seem to exist in the DNS
    aren't getting matched by the system or are returned with a null string
    result. This is due to the NAPTR reply requiring a "+" in the regular
    expression matching sequence. Older versions of Asterisk add a "+"
    from within the code, which may confuse administrators converting to
    the new function. Please ensure that all ENUM (e164.arpa) lookups
    contain a leading "+" before lookup, so ensure your lookup includes
    the leading plus sign. Other DNS trees may or may not require a lead-
    ing "+" - check before using those trees, as it is possible the parsed
    NAPTRs will not provide correct results unless you have the correct
    dialed string. If you get console messages like "WARNING[24907]:
    enum.c:222 parse naptr: NAPTR Regex match failed." then it is very
    possible that the returned NAPTR expects a leading "+" in the search
    string (or the returned NAPTR is mis-formed.)

  * If a query is performed of type "c" ("count") and let's say you get
    back 5 records and then some seconds later a query is made against
    record 5 in the list, it may not be the case that the DNS resolver has
    the same answers as it did a second or two ago - maybe there are only
    4 records in the list in the newest query. The resolver should be the
    canonical storage location for DNS records, since that is the intent of
    ENUM. However, some obscure future cases may have wildly changing
    NAPTR records within several seconds. This is a corner case, and
    probably only worth noting as a very rare circumstance. (note: I do
                                         136


  not object to Asterisk's dnsmgr method of locally caching DNS replies,
  but this method needs to honor the TTL given by the remote zone
  master. Currently, the ENUMLOOKUP function does not use the
  dnsmgr method of caching local DNS replies.)
* If you want strict NAPTR value ordering, then it will be necessary
  to use the "ALL" method to incrementally step through the different
  returned NAPTR pointers. You will need to use string manipulation
  to strip off the returned method types, since the results will look like
  "sip:12125551212" in the returned value. This is a non-trivial task,
  though it is required in order to have strict RFC compliance and to
  comply with the desires of the remote party who is presenting NAPTRs
  in a particular order for a reason.

* Default behavior for the function (even in event of an error) is to move
  to the next priority, and the result is a null value. Most ENUM lookups
  are going to be failures, and it is the responsibility of the dialplan
  administrator to manage error conditions within their dialplan. This
  is a change from the old app enumlookup method and it's arbitrary
  priority jumping based on result type or failure.
* Anything other than digits will be ignored in lookup strings. Example:
  a search string of "+4372030blah01721" will turn into 1.2.7.1.0.0.3.0.2.7.3.4.e164.arpa.
  for the lookup. The NAPTR parsing may cause unexpected results if
  there are strings inside your NAPTR lookups.
* If there exist multiple records with the same weight and order as a result
  of your query, the function will RANDOMLY select a single NAPTR
  from those equal results.
* Currently, the function ignores the settings in enum.conf as the search
  zone name is now specified within the function, and the H323 driver
  can be chosen by the user via the dialplan. There were no other values
  in this file, and so it becomes deprecated.

* The function will digest and return NAPTRs which use older (depre-
  cated) style, reversed method strings such as "sip+E2U" instead of the
  more modern "E2U+sip"


                                       137


   * There is no provision for multi-part methods at this time. If there are
      multiple NAPTRs with (as an example) a method of "E2U+voice:sip"
      and then another NAPTR in the same DNS record with a method of
      ""E2U+sip", the system will treat these both as method "sip" and they
      will be separate records from the perspective of the function. Of course,
      if both records point to the same URI and have equal priority/weight
      (as is often the case) then this will cause no serious difficulty, but it
      bears mentioning.
   * ISN (ITAD Subscriber Number) usage: If the search number is of the
      form ABC*DEF (where ABC and DEF are at least one numeric digit)
      then perform an ISN-style lookup where the lookup is manipulated
      to C.B.A.DEF.domain.tld (all other settings and options apply.) See
      http://www.freenum.org/ for more details on ISN lookups. In the un-
      likely event you wish to avoid ISN re-writes, put an "n" as the first
      digit of the search string - the "n" will be ignored for the search.

8.1.4 Some more Examples

All examples below except where noted use "e164.arpa" as the referenced
domain, which is the default domain name for ENUMLOOKUP. All numbers
are assumed to not have a leading "+" as dialed by the inbound channel,
so that character is added where necessary during ENUMLOOKUP function
calls.
  ; example 1
  ;
  ; Assumes North American international dialing (011) prefix.
  ; Look up the first SIP result and send the call there, otherwise
  ; send the call out a PRI. This is the most simple possible
  ; ENUM example, but only uses the first SIP reply in the list of
  ; NAPTR(s).
  ;
  exten => _011.,1,Set(enumresult=${ENUMLOOKUP(+${EXTEN:3})})
  exten => _011.,n,Dial(SIP/${enumresult})
  exten => _011.,n,Dial(DAHDI/g1/${EXTEN})
  ;
  ; end example 1
  ; example 2
  ;
  ; Assumes North American international dialing (011) prefix.
  ; Check to see if there are multiple SIP NAPTRs returned by
  ; the lookup, and dial each in order. If none work (or none
  ; exist) then send the call out a PRI, group 1.

                                            138


;
exten => _011.,1,Set(sipcount=${ENUMLOOKUP(${EXTEN:3},sip,c)}|counter=0)
exten => _011.,n,While($["${counter}"<"${sipcount}"])
exten => _011.,n,Set(counter=$[${counter}+1])
exten => _011.,n,Dial(SIP/${ENUMLOOKUP(+${EXTEN:3},sip,,${counter})})
exten => _011.,n,EndWhile
exten => _011.,n,Dial(DAHDI/g1/${EXTEN})
;
; end example 2
; example 3
;
; This example expects an ${EXTEN} that is an e.164 number (like
; 14102241145 or 437203001721)
; Search through e164.arpa and then also search through e164.org
; to see if there are any valid SIP or IAX termination capabilities.
; If none, send call out via DAHDI channel 1.
;
; Start first with e164.arpa zone...
;
exten => _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten => _X.,2,GotoIf($["${counter}"<"${sipcount}"]?3:6)
exten => _X.,3,Set(counter=$[${counter}+1])
exten => _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,,${counter})})
exten => _X.,5,GotoIf($["${counter}"<"${sipcount}"]?3:6)
;
exten => _X.,6,Set(iaxcount=${ENUMLOOKUP(+${EXTEN},iax2,c)}|counter=0)
exten => _X.,7,GotoIf($["${counter}"<"${iaxcount}"]?8:11)
exten => _X.,8,Set(counter=$[${counter}+1])
exten => _X.,9,Dial(IAX2/${ENUMLOOKUP(+${EXTEN},iax2,,${counter})})
exten => _X.,10,GotoIf($["${counter}"<"${iaxcount}"]?8:11)
;
exten => _X.,11,NoOp("No valid entries in e164.arpa for ${EXTEN} - checking in e164.org")
;
; ...then also try e164.org, and look for SIP and IAX NAPTRs...
;
exten => _X.,12,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c,,e164.org)}|counter=0)
exten => _X.,13,GotoIf($["${counter}"<"${sipcount}"]?14:17)
exten => _X.,14,Set(counter=$[${counter}+1])
exten => _X.,15,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,,${counter},e164.org)})
exten => _X.,16,GotoIf($["${counter}"<"${sipcount}"]?14:17)
;
exten => _X.,17,Set(iaxcount=${ENUMLOOKUP(+${EXTEN},iax2,c,,e164.org)}|counter=0)
exten => _X.,18,GotoIf($["${counter}"<"${iaxcount}"]?19:22)
exten => _X.,19,Set(counter=$[${counter}+1])
exten => _X.,20,Dial(IAX2/${ENUMLOOKUP(+${EXTEN},iax2,,${counter},e164.org)})
exten => _X.,21,GotoIf($["${counter}"<"${iaxcount}"]?19:22)
;
; ...then send out PRI.
;
exten => _X.,22,NoOp("No valid entries in e164.org for ${EXTEN} - sending out via DAHDI")
exten => _X.,23,Dial(DAHDI/g1/${EXTEN})
;
; end example 3




                                         139




Chapter 9

AMI: Asterisk Manager Interface


9.1 The Asterisk Manager TCP/IP API

The manager is a client/server model over TCP. With the manager interface,
you'll be able to control the PBX, originate calls, check mailbox status,
monitor channels and queues as well as execute Asterisk commands.
    AMI is the standard management interface into your Asterisk server. You
configure AMI in manager.conf. By default, AMI is available on TCP port
5038 if you enable it in manager.conf.
    AMI receive commands, called "actions". These generate a "response"
from Asterisk. Asterisk will also send "Events" containing various infor-
mation messages about changes within Asterisk. Some actions generate an
initial response and data in the form list of events. This format is created to
make sure that extensive reports do not block the manager interface fully.
    Management users are configured in the configuration file manager.conf
and are given permissions for read and write, where write represents their
ability to perform this class of "action", and read represents their ability to
receive this class of "event".
    If you develop AMI applications, treat the headers in Actions, Events
and Responses as local to that particular message. There is no cross-message
standardization of headers.
    If you develop applications, please try to reuse existing manager head-
ers and their interpretation. If you are unsure, discuss on the asterisk-dev
mailing list.
    Manager subscribes to extension status reports from all channels, to be
                                           140


able to generate events when an extension or device changes state. The level
of details in these events may depend on the channel and device configura-
tion. Please check each channel configuration file for more information. (in
sip.conf, check the section on subscriptions and call limits)

9.2 Command Syntax
Management communication consists of tags of the form "header: value",
terminated with an empty newline (\r\n) in the style of SMTP, HTTP, and
other headers.
   The first tag MUST be one of the following:

   * Action: An action requested by the CLIENT to the Asterisk SERVER.
      Only one "Action" may be outstanding at any time.
   * Response: A response to an action from the Asterisk SERVER to the
      CLIENT.
   * Event: An event reported by the Asterisk SERVER to the CLIENT

9.3 Manager commands

To see all of the available manager commands, use the "manager show com-
mands" CLI command.
   You can get more information about a manager command with the "man-
ager show command <command>" CLI command in Asterisk.

9.4 Examples
Login - Log a user into the manager interface.

    Action: Login
    Username: testuser
    Secret: testsecret
   Originate - Originate a call from a channel to an extension.


                                         141


    Action: Originate
    Channel: sip/12345
    Exten: 1234
    Context: default
    Originate - Originate a call from a channel to an extension without waiting
for call to complete.
    Action: Originate
    Channel: sip/12345
    Exten: 1234
    Context: default
    Async: yes
    Redirect with ExtraChannel:
    Attempted goal: Have a 'robot' program Redirect both ends of an already-
connected call to a meetme room using the ExtraChannel feature through
the management interface.
   Action: Redirect
   Channel: DAHDI/1-1
   ExtraChannel: SIP/3064-7e00 (varies)
   Exten: 680
   Priority: 1
    Where 680 is an extension that sends you to a MeetMe room.
    There are a number of GUI tools that use the manager interface, please
search the mailing list archives and the documentation page on the http://www.asterisk.org
web site for more information.

9.5 Ensuring all modules are loaded

It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that
might require a module that has not yet loaded, the application can listen
for the FullyBooted manager event. It will be sent upon connection if all
modules have been loaded, or as soon as loading is complete. The event:
    Event: FullyBooted
    Privilege: system,all
    Status: Fully Booted
                                            142


9.6 Device status reports

9.7 Some standard AMI headers
Account: -- Account Code (Status)
AccountCode: -- Account Code (cdr_manager)
ACL: <Y | N> -- Does ACL exist for object ?
Action: <action> -- Request or notification of a particular action
Address-IP: -- IPaddress
Address-Port: -- IP port number
Agent: <string> -- Agent name
AMAflags: -- AMA flag (cdr_manager, sippeers)
AnswerTime: -- Time of answer (cdr_manager)
Append: <bool> -- CDR userfield Append flag
Application: -- Application to use
Async: -- Whether or not to use fast setup
AuthType: -- Authentication type (for login or challenge)
         "md5"
BillableSeconds: -- Billable seconds for call (cdr_manager)
CallerID: -- Caller id (name and number in Originate & cdr_manager)
CallerID: -- CallerID number
                            Number or "<unknown>" or "unknown"
                            (should change to "<unknown>" in app_queue)
CallerID1: -- Channel 1 CallerID (Link event)
CallerID2: -- Channel 2 CallerID (Link event)
CallerIDName: -- CallerID name
                            Name or "<unknown>" or "unknown"
                            (should change to "<unknown>" in app_queue)
Callgroup: -- Call group for peer/user
CallsTaken: <num> -- Queue status variable
Cause: <value> -- Event change cause - "Expired"
Cause: <value> -- Hangupcause (channel.c)
CID-CallingPres: -- Caller ID calling presentation
Channel: <channel> -- Channel specifier
Channel: <dialstring> -- Dialstring in Originate
Channel: <tech/[peer/username]> -- Channel in Registry events (SIP, IAX2)
Channel: <tech> -- Technology (SIP/IAX2 etc) in Registry events


                             143


ChannelType: -- Tech: SIP, IAX2, DAHDI, MGCP etc
Channel1: -- Link channel 1
Channel2: -- Link channel 2
ChanObjectType: -- "peer", "user"
Codecs: -- Codec list
CodecOrder: -- Codec order, separated with comma ","
Command: -- Cli command to run
Context: -- Context
Count: <num> -- Number of callers in queue
Data: -- Application data
Default-addr-IP: -- IP address to use before registration
Default-Username: -- Username part of URI to use before registration
Destination: -- Destination for call (Dialstring ) (dial, cdr_manager)
DestinationContext: -- Destination context (cdr_manager)
DestinationChannel: -- Destination channel (cdr_manager)
DestUniqueID: -- UniqueID of destination (dial event)
Disposition: -- Call disposition (CDR manager)
Domain: <domain> -- DNS domain
Duration: <secs> -- Duration of call (cdr_manager)
Dynamic: <Y | N> -- Device registration supported?
Endtime: -- End time stamp of call (cdr_manager)
EventList: <flag> -- Flag being "Start", "End", "Cancelled" or "ListObject"
Events: <eventmask> -- Eventmask filter ("on", "off", "system", "call", "log")
Exten: -- Extension (Redirect command)
Extension: -- Extension (Status)
Family: <string> -- ASTdb key family
File: <filename> -- Filename (monitor)
Format: <format> -- Format of sound file (monitor)
From: <time> -- Parking time (ParkedCall event)
Hint: -- Extension hint
Incominglimit: -- SIP Peer incoming limit
Key:
Key: -- ASTdb Database key
LastApplication: -- Last application executed (cdr_manager)
LastCall: <num> -- Last call in queue
LastData: -- Data for last application (cdr_manager)
Link: -- (Status)
ListItems: <number> -- Number of items in Eventlist (Optionally sent in "end" packet)

                          144


Location: -- Interface (whatever that is -maybe tech/name in app_queue )
Loginchan: -- Login channel for agent
Logintime: <number> -- Login time for agent
Mailbox: -- VM Mailbox (id@vmcontext) (mailboxstatus, mailboxcount)
MD5SecretExist: <Y | N> -- Whether secret exists in MD5 format
Membership: <string> -- "Dynamic" or "static" member in queue
Message: <text> -- Text message in ACKs, errors (explanation)
Mix: <bool> -- Boolean parameter (monitor)
NewMessages: <count> -- Count of new Mailbox messages (mailboxcount)
Newname:
ObjectName: -- Name of object in list
OldName: -- Something in Rename (channel.c)
OldMessages: <count> -- Count of old mailbox messages (mailboxcount)
Outgoinglimit: -- SIP Peer outgoing limit
Paused: <num> -- Queue member paused status
Peer: <tech/name> -- "channel" specifier :-)
PeerStatus: <tech/name> -- Peer status code
                         "Unregistered", "Registered", "Lagged", "Reachable"
Penalty: <num> -- Queue penalty
Priority: -- Extension priority
Privilege: <privilege> -- AMI authorization class (system, call, log, verbose, command, agent, user)
Pickupgroup: -- Pickup group for peer
Position: <num> -- Position in Queue
Queue: -- Queue name
Reason: -- "Autologoff"
Reason: -- "Chanunavail"
Response: <response> -- response code, like "200 OK"
                         "Success", "Error", "Follows"
Restart: -- "True", "False"
RegExpire: -- SIP registry expire
RegExpiry: -- SIP registry expiry
Reason: -- Originate reason code
Seconds: -- Seconds (Status)
Secret: <password> -- Authentication secret (for login)
SecretExist: <Y | N> -- Whether secret exists
Shutdown: -- "Uncleanly", "Cleanly"
SIP-AuthInsecure:
SIP-FromDomain: -- Peer FromDomain

                            145


 SIP-FromUser: -- Peer FromUser
 SIP-NatSupport:
 SIPLastMsg:
 Source: -- Source of call (dial event, cdr_manager)
 SrcUniqueID: -- UniqueID of source (dial event)
 StartTime: -- Start time of call (cdr_manager)
 State: -- Channel state
 Status: -- Registration status (Registry events SIP)
 Status: -- Extension status (Extensionstate)
 Status: -- Peer status (if monitored) ** Will change name **
                                    "unknown", "lagged", "ok"
 Status: <num> -- Queue Status
 Status: -- DND status (DNDState)
 Time: <sec> -- Roundtrip time (latency)
 Timeout: -- Parking timeout time
 Timeout: -- Timeout for call setup (Originate)
 Timeout: <seconds> -- Timeout for call
 Uniqueid: -- Channel Unique ID
 Uniqueid1: -- Channel 1 Unique ID (Link event)
 Uniqueid2: -- Channel 2 Unique ID (Link event)
 User: -- Username (SIP registry)
 UserField: -- CDR userfield (cdr_manager)
 Val: -- Value to set/read in ASTdb
 Variable: -- Variable AND value to set (multiple separated with | in Originate)
 Variable: <name> -- For channel variables
 Value: <value> -- Value to set
 VoiceMailbox: -- VM Mailbox in SIPpeers
 Waiting: -- Count of mailbox messages (mailboxstatus)
   ** Please try to re-use existing headers to simplify manager message
parsing in clients.
   Read the CODING-GUIDELINES if you develop new manager commands
or events.

9.8 Asynchronous Javascript Asterisk Manger (AJAM)
AJAM is a new technology which allows web browsers or other HTTP enabled
applications and web pages to directly access the Asterisk Manger Interface

                                      146


(AMI) via HTTP. Setting up your server to process AJAM involves a few
steps:

9.8.1 Setup the Asterisk HTTP server
   1. Uncomment the line "enabled=yes" in /etc/asterisk/http.conf to
      enable Asterisk's builtin micro HTTP server.
   2. If you want Asterisk to actually deliver simple HTML pages, CSS,
      javascript, etc. you should uncomment "enablestatic=yes"

   3. Adjust your "bindaddr" and "bindport" settings as appropriate for
      your desired accessibility
   4. Adjust your "prefix" if appropriate, which must be the beginning of
      any URI on the server to match. The default is "asterisk" and the rest
      of these instructions assume that value.

9.8.2 Allow Manager Access via HTTP
   1. Make sure you have both "enabled = yes" and "webenabled = yes"
      setup in /etc/asterisk/manager.conf
   2. You may also use "httptimeout" to set a default timeout for HTTP
      connections.

   3. Make sure you have a manager username/secret
    Once those configurations are complete you can reload or restart Asterisk
and you should be able to point your web browser to specific URI's which
will allow you to access various web functions. A complete list can be found
by typing "http show status" at the Asterisk CLI.
    examples:
  http://localhost:8088/asterisk/manager?action=login&username=foo&secret=bar

This logs you into the manager interface's "HTML" view. Once you're logged
in, Asterisk stores a cookie on your browser (valid for the length of httpti-
meout) which is used to connect to the same session.
  http://localhost:8088/asterisk/rawman?action=status

                                           147


Assuming you've already logged into manager, this URI will give you a "raw"
manager output for the "status" command.
   http://localhost:8088/asterisk/mxml?action=status
This will give you the same status view but represented as AJAX data,
theoretically compatible with RICO (http://www.openrico.org).
   http://localhost:8088/asterisk/static/ajamdemo.html

If you have enabled static content support and have done a make install, As-
terisk will serve up a demo page which presents a live, but very basic, "ast-
man" like interface. You can login with your username/secret for manager
and have a basic view of channels as well as transfer and hangup calls. It's
only tested in Firefox, but could probably be made to run in other browsers
as well.
    A sample library (astman.js) is included to help ease the creation of man-
ager HTML interfaces.
    Note that for the demo, there is no need for *any* external web server.

9.8.3 Integration with other web servers
Asterisk's micro HTTP server is *not* designed to replace a general purpose
web server and it is intentionally created to provide only the minimal inter-
faces required. Even without the addition of an external web server, one can
use Asterisk's interfaces to implement screen pops and similar tools pulling
data from other web servers using iframes, div's etc. If you want to integrate
CGI's, databases, PHP, etc. you will likely need to use a more traditional
web server like Apache and link in your Asterisk micro HTTP server with
something like this:
   ProxyPass /asterisk http://localhost:8088/asterisk








                                            148



Chapter 10


CDR: Call Detail Records


10.1 Applications
   * SetAccount - Set account code for billing
   * SetAMAFlags - Sets AMA flags
   * NoCDR - Make sure no CDR is saved for a specific call
   * ResetCDR - Reset CDR

   * ForkCDR - Save current CDR and start a new CDR for this call
   * Authenticate - Authenticates and sets the account code
   * SetCDRUserField - Set CDR user field
   * AppendCDRUserField - Append data to CDR User field
   For more information, use the "core show application <application>"
command. You can set default account codes and AMA flags for devices in
channel configuration files, like sip.conf, iax.conf etc.

10.2 CDR Fields

   * accountcode: What account number to use, (string, 20 characters)

                                        149


* src: Caller*ID number (string, 80 characters)
* dst: Destination extension (string, 80 characters)
* dcontext: Destination context (string, 80 characters)

* clid: Caller*ID with text (80 characters)
* channel: Channel used (80 characters)
* dstchannel: Destination channel if appropriate (80 characters)
* lastapp: Last application if appropriate (80 characters)

* lastdata: Last application data (arguments) (80 characters)
* start: Start of call (date/time)
* answer: Answer of call (date/time)
* end: End of call (date/time)

* duration: Total time in system, in seconds (integer), from dial to
  hangup
* billsec: Total time call is up, in seconds (integer), from answer to
  hangup
* disposition: What happened to the call: ANSWERED, NO ANSWER,
  BUSY

* amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE
  etc, specified on a per channel basis like accountcode.
* user field: A user-defined field, maximum 255 characters
In some cases, uniqueid is appended:

* uniqueid: Unique Channel Identifier (32 characters) This needs to be
  enabled in the source code at compile time


                                       150


    NOTE: If you use IAX2 channels for your calls, and allow 'full' transfers
(not media-only transfers), then when the calls is transferred the server in
the middle will no longer be involved in the signaling path, and thus will not
generate accurate CDRs for that call. If you can, use media-only transfers
with IAX2 to avoid this problem, or turn off transfers completely (although
this can result in a media latency increase since the media packets have to
traverse the middle server(s) in the call).

10.3 Variables
If the channel has a CDR, that CDR has its own set of variables which can
be accessed just like channel variables. The following builtin variables are
available.

${CDR(clid)} Caller ID
${CDR(src)} Source
${CDR(dst)} Destination
${CDR(dcontext)} Destination context
${CDR(channel)} Channel name
${CDR(dstchannel)} Destination channel
${CDR(lastapp)} Last app executed
${CDR(lastdata)} Last app's arguments
${CDR(start)} Time the call started.
${CDR(answer)} Time the call was answered.
${CDR(end)} Time the call ended.
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
${CDR(accountcode)} The channel's account code.
${CDR(uniqueid)} The channel's unique id.
${CDR(userfield)} The channels uses specified field.
    In addition, you can set your own extra variables by using Set(CDR(name)=value).
These variables can be output into a text-format CDR by using the cdr cus-
tom CDR driver; see the cdr custom.conf.sample file in the configs directory
for an example of how to do this.

                                         151


10.4 Storage Backends

10.4.1 Microsoft SQL Server
Asterisk can currently store CDRs into an MSSQL database in two different
ways: cdr odbc or cdr tds
   Call Data Records can be stored using unixODBC (which requires the
FreeTDS package) [cdr odbc] or directly by using just the FreeTDS package
[cdr tds] The following provide some examples known to get asterisk working
with mssql.
   NOTE: Only choose one db connector.

ODBC using cdr odbc

Compile, configure, and install the latest unixODBC package:
  tar -zxvf unixODBC-2.2.9.tar.gz &&
  cd unixODBC-2.2.9 &&
  ./configure --sysconfdir=/etc --prefix=/usr --disable-gui &&
  make &&
  make install
   Compile, configure, and install the latest FreeTDS package:
  tar -zxvf freetds-0.62.4.tar.gz &&
  cd freetds-0.62.4 &&
  ./configure --prefix=/usr --with-tdsver=7.0 \
                   --with-unixodbc=/usr/lib &&
  make && make install

   Compile, or recompile, asterisk so that it will now add support for cdr -
odbc.
  make clean && ./configure --with-odbc &&
  make update &&
  make &&
  make install
   Setup odbc configuration files. These are working examples from my
system. You will need to modify for your setup. You are not required to
store usernames or passwords here.
  /etc/odbcinst.ini
     [FreeTDS]
     Description = FreeTDS ODBC driver for MSSQL
     Driver = /usr/lib/libtdsodbc.so
     Setup = /usr/lib/libtdsS.so
     FileUsage = 1

                                           152


  /etc/odbc.ini
     [MSSQL-asterisk]
     description = Asterisk ODBC for MSSQL
     driver = FreeTDS
     server = 192.168.1.25
     port = 1433
     database = voipdb
     tds_version = 7.0
     language = us_english
   Only install one database connector. Do not confuse asterisk by using
both ODBC (cdr odbc) and FreeTDS (cdr tds). This command will erase
the contents of cdr tds.conf
  [ -f /etc/asterisk/cdr_tds.conf ] > /etc/asterisk/cdr_tds.conf
NOTE: unixODBC requires the freeTDS package, but asterisk does not call
freeTDS directly.
   Now set up cdr odbc configuration files. These are working samples from
my system. You will need to modify for your setup. Define your usernames
and passwords here, secure file as well.
  /etc/asterisk/cdr_odbc.conf
     [global]
     dsn=MSSQL-asterisk
     username=voipdbuser
     password=voipdbpass
     loguniqueid=yes

And finally, create the 'cdr' table in your mssql database.
  CREATE TABLE cdr (
          [calldate] [datetime] NOT NULL ,
          [clid] [varchar] (80) NOT NULL ,
          [src] [varchar] (80) NOT NULL ,
          [dst] [varchar] (80) NOT NULL ,
          [dcontext] [varchar] (80) NOT NULL ,
          [channel] [varchar] (80) NOT NULL ,
          [dstchannel] [varchar] (80) NOT NULL ,
          [lastapp] [varchar] (80) NOT NULL ,
          [lastdata] [varchar] (80) NOT NULL ,
          [duration] [int] NOT NULL ,
          [billsec] [int] NOT NULL ,
          [disposition] [varchar] (45) NOT NULL ,
          [amaflags] [int] NOT NULL ,
          [accountcode] [varchar] (20) NOT NULL ,
          [uniqueid] [varchar] (150) NOT NULL ,
          [userfield] [varchar] (255) NOT NULL
  )
Start asterisk in verbose mode, you should see that asterisk logs a connection
to the database and will now record every call to the database when it's
complete.

                                            153


TDS, using cdr tds

Compile, configure, and install the latest FreeTDS package:
     tar -zxvf freetds-0.62.4.tar.gz &&
     cd freetds-0.62.4 &&
     ./configure --prefix=/usr --with-tdsver=7.0
     make &&
     make install
Compile, or recompile, asterisk so that it will now add support for cdr tds.

                     make clean && ./configure --with-tds &&
                     make update &&
                     make &&
                     make install
Only install one database connector. Do not confuse asterisk by using both
ODBC (cdr odbc) and FreeTDS (cdr tds). This command will erase the
contents of cdr odbc.conf
  [ -f /etc/asterisk/cdr_odbc.conf ] > /etc/asterisk/cdr_odbc.conf

Setup cdr tds configuration files. These are working samples from my system.
You will need to modify for your setup. Define your usernames and passwords
here, secure file as well.
                  /etc/asterisk/cdr_tds.conf
     [global]
     hostname=192.168.1.25
     port=1433
     dbname=voipdb
     user=voipdbuser
     password=voipdpass
     charset=BIG5

And finally, create the 'cdr' table in your mssql database.
  CREATE TABLE cdr (
          [accountcode] [varchar] (20) NULL ,
          [src] [varchar] (80) NULL ,
          [dst] [varchar] (80) NULL ,
          [dcontext] [varchar] (80) NULL ,
          [clid] [varchar] (80) NULL ,
          [channel] [varchar] (80) NULL ,
          [dstchannel] [varchar] (80) NULL ,
          [lastapp] [varchar] (80) NULL ,
          [lastdata] [varchar] (80) NULL ,
          [start] [datetime] NULL ,
          [answer] [datetime] NULL ,
          [end] [datetime] NULL ,
          [duration] [int] NULL ,
          [billsec] [int] NULL ,

                                            154


          [disposition] [varchar] (20) NULL ,
          [amaflags] [varchar] (16) NULL ,
          [uniqueid] [varchar] (150) NULL ,
          [userfield] [varchar] (256) NULL
  )
Start asterisk in verbose mode, you should see that asterisk logs a connection
to the database and will now record every call to the database when it's
complete.


10.4.2 MySQL
Using MySQL for CDR records is supported by using ODBC and the cdr -
odbc module.

10.4.3 PostgreSQL
If you want to go directly to postgresql database, and have the cdr pgsql.so
compiled you can use the following sample setup. On Debian, before compil-
ing asterisk, just install libpqxx-dev. Other distros will likely have a similiar
package.
    Once you have the compile done, copy the sample cdr pgsql.conf file or
create your own.
    Here is a sample:

          /etc/asterisk/cdr_pgsql.conf
            ; Sample Asterisk config file for CDR logging to PostgresSQL
            [global]
            hostname=localhost
            port=5432
            dbname=asterisk
            password=password
            user=postgres
            table=cdr
Now create a table in postgresql for your cdrs
          CREATE TABLE cdr (
                  calldate time NOT NULL ,
                  clid varchar (80) NOT NULL ,
                  src varchar (80) NOT NULL ,
                  dst varchar (80) NOT NULL ,
                  dcontext varchar (80) NOT NULL ,
                  channel varchar (80) NOT NULL ,
                  dstchannel varchar (80) NOT NULL ,
                  lastapp varchar (80) NOT NULL ,
                  lastdata varchar (80) NOT NULL ,

                                           155


                duration int NOT NULL ,
                billsec int NOT NULL ,
                disposition varchar (45) NOT NULL ,
                amaflags int NOT NULL ,
                accountcode varchar (20) NOT NULL ,
                uniqueid varchar (150) NOT NULL ,
                userfield varchar (255) NOT NULL
         );

10.4.4 SQLite 2
SQLite version 2 is supported in cdr sqlite.

10.4.5 SQLite 3
SQLite version 3 is supported in cdr sqlite3 custom.

10.4.6 RADIUS
What is needed
   * FreeRADIUS server

   * Radiusclient-ng library
   * Asterisk PBX


               Asterisk PBX

                                              - RADIUS Server
              RADIUS Client
                                       oe (FreeRADIUS)


                 Figure 10.1: Asterisk/RADIUS Integration





                                       156


Installation of the Radiusclient library
Download the sources from http://developer.berlios.de/projects/radiusclient-ng/
    Untar the source tarball:

root@localhost:/usr/local/src# tar xvfz radiusclient-ng-0.5.2.tar.gz
    Compile and install the library:

root@localhost:/usr/local/src# cd radiusclient-ng-0.5.2
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# ./configure
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make install
Configuration of the Radiusclient library
By default all the configuration files of the radiusclient library will be in
/usr/local/etc/radiusclient-ng directory.
    File "radiusclient.conf" Open the file and find lines containing the follow-
ing:
    authserver localhost
    This is the hostname or IP address of the RADIUS server used for au-
thentication. You will have to change this unless the server is running on the
same host as your Asterisk PBX.
    acctserver localhost
    This is the hostname or IP address of the RADIUS server used for ac-
counting. You will have to change this unless the server is running on the
same host as your Asterisk PBX.
    File "servers"
    RADIUS protocol uses simple access control mechanism based on shared
secrets that allows RADIUS servers to limit access from RADIUS clients.
    A RADIUS server is configured with a secret string and only RADIUS
clients that have the same secret will be accepted.
    You need to configure a shared secret for each server you have configured
in radiusclient.conf file in the previous step. The shared secrets are stored in
/usr/local/etc/radiusclient-ng/servers file.
    Each line contains hostname of a RADIUS server and shared secret used
in communication with that server. The two values are separated by white


                                           157


spaces. Configure shared secrets for every RADIUS server you are going to
use.
    File "dictionary"
    Asterisk uses some attributes that are not included in the dictionary of
radiusclient library, therefore it is necessary to add them. A file called dic-
tionary.digium (kept in the contrib dir) was created to list all new attributes
used by Asterisk. Add to the end of the main dictionary file /usr/local/
etc/radiusclient-ng/dictionary the line:
    $INCLUDE /path/to/dictionary.digium

Install FreeRADIUS Server (Version 1.1.1)
Download sources tarball from:
    http://freeradius.org/
    Untar, configure, build, and install the server:
root@localhost:/usr/local/src# tar xvfz freeradius-1.1.1.tar.gz
root@localhost:/usr/local/src# cd freeradius-1.1.1
root@localhost"/usr/local/src/freeradius-1.1.1# ./configure
root@localhost"/usr/local/src/freeradius-1.1.1# make
root@localhost"/usr/local/src/freeradius-1.1.1# make install
    All the configuration files of FreeRADIUSserver will be in /usr/local/etc/raddb
directory.

Configuration of the FreeRADIUS Server
There are several files that have to be modified to configure the RADIUS
server. These are presented next.
    File "clients.conf"
    File /usr/local/etc/raddb/clients.conf contains description of RA-
DIUS clients that are allowed to use the server. For each of the clients you
need to specify its hostname or IP address and also a shared secret. The
shared secret must be the same string you configured in radiusclient library.
    Example:
client myhost {
      secret = mysecret
      shortname = foo
}

                                             158


    This fragment allows access from RADIUS clients on "myhost" if they
use "mysecret" as the shared secret. The file already contains an entry for
localhost (127.0.0.1), so if you are running the RADIUS server on the same
host as your Asterisk server, then modify the existing entry instead, replacing
the default password.
    File "dictionary"
    Note: as of version 1.1.2, the dictionary.digium file ships with FreeRA-
DIUS. The following procedure brings the dictionary.digium file to previous
versions of FreeRADIUS.
    File /usr/local/etc/raddb/dictionary contains the dictionary of FreeRA-
DIUS server. You have to add the same dictionary file (dictionary.digium),
which you added to the dictionary of radiusclient-ng library. You can include
it into the main file, adding the following line at the end of file /usr/local/
etc/raddb/dictionary:
    $INCLUDE /path/to/dictionary.digium
    That will include the same new attribute definitions that are used in
radiusclient-ng library so the client and server will understand each other.

Asterisk Accounting Configuration
Compilation and installation:
    The module will be compiled as long as the radiusclient-ng library has
been detected on your system.
    By default FreeRADIUS server will log all accounting requests into /usr/
local/var/log/radius/radacct directory in form of plain text files. The
server will create one file for each hostname in the directory. The following
example shows how the log files look like.
    Asterisk now generates Call Detail Records. See /include/asterisk/
cdr.h for all the fields which are recorded. By default, records in comma
separated values will be created in /var/log/asterisk/cdr-csv.
    The configuration file for cdr radius.so module is /etc/asterisk/cdr.
conf
    This is where you can set CDR related parameters as well as the path to
the radiusclient-ng library configuration file.
Logged Values

   "Asterisk-Acc-Code", The account name of detail records

                                             159


"Asterisk-Src",
"Asterisk-Dst",
"Asterisk-Dst-Ctx", The destination context
"Asterisk-Clid",
"Asterisk-Chan", The channel
"Asterisk-Dst-Chan", (if applicable)
"Asterisk-Last-App", Last application run on the channel
"Asterisk-Last-Data", Argument to the last channel
"Asterisk-Start-Time",
"Asterisk-Answer-Time",
"Asterisk-End-Time",
"Asterisk-Duration", Duration is the whole length that the entire
                           call lasted. ie. call rx'd to hangup
                           "end time" minus "start time"
"Asterisk-Bill-Sec", The duration that a call was up after other
                           end answered which will be <= to duration
                           "end time" minus "answer time"
"Asterisk-Disposition", ANSWERED, NO ANSWER, BUSY
"Asterisk-AMA-Flags", DOCUMENTATION, BILL, IGNORE etc, specified on
                           a per channel basis like accountcode.
"Asterisk-Unique-ID", Unique call identifier
"Asterisk-User-Field" User field set via SetCDRUserField













                         160



Chapter 11


Voicemail


11.1 ODBC Storage
ODBC Storage allows you to store voicemail messages within a database
instead of using a file. This is not a full realtime engine and only supports
ODBC. The table description for the voicemessages table is as follows:
       Field Type Null Key Default Extra
       msgnum int(11) Yes NULL
       dir varchar(80) Yes MUL NULL
       context varchar(80) Yes NULL
       macrocontext varchar(80) Yes NULL
       callerid varchar(40) Yes NULL
       origtime varchar(40) Yes NULL
       duration varchar(20) Yes NULL
       flag varchar(8) Yes NULL
       mailboxuser varchar(80) Yes NULL
       mailboxcontext varchar(80) Yes NULL
       recording longblob Yes NULL

                 Table 11.1: voicemessages Table Schema

   The database name (from /etc/asterisk/res_odbc.conf) is in the odbcstorage
variable in the general section of voicemail.conf.
                                        161


    You may modify the voicemessages table name by using odbctable=table_-
name in voicemail.conf.

11.2 IMAP Storage
By enabling IMAP Storage, Asterisk will use native IMAP as the storage
mechanism for voicemail messages instead of using the standard file structure.
    Tighter integration of Asterisk voicemail and IMAP email services allows
additional voicemail functionality, including:

    * Listening to a voicemail on the phone will set its state to "read" in a
      user's mailbox automatically.
    * Deleting a voicemail on the phone will delete it from the user's mailbox
      automatically.
    * Accessing a voicemail recording email message will turn off the message
      waiting indicator (MWI) on the user's phone.

    * Deleting a voicemail recording email will also turn off the message
      waiting indicator, and delete the message from the voicemail system.
11.2.1 Installation Notes

University of Washington IMAP C-Client
If you do not have the University of Washington's IMAP c-client installed
on your system, you will need to download the c-client source distribution
(http://www.washington.edu/imap/) and compile it. Asterisk supports the
2007 version of c-client as there appears to be issues with older versions which
cause Asterisk to crash in certain scenarios. It is highly recommended that
you utilize a current version of the c-client libraries. Additionally, mail -
expunge full is enabled in the 2006 and later versions.
    Note that Asterisk only uses the 'c-client' portion of the UW IMAP
toolkit, but building it also builds an IMAP server and various other utilities.
Because of this, the build instructions for the IMAP toolkit are somewhat
complicated and can lead to confusion about what is needed.
    If you are going to be connecting Asterisk to an existing IMAP server,
then you don't need to care about the server or utilities in the IMAP toolkit
                                          162


at all. If you want to also install the UW IMAPD server, that is outside the
scope of this document.
   Building the c-client library is fairly straightforward; for example, on a
Debian system there are two possibilities:
   1. If you will not be using SSL to connect to the IMAP server:

                  $ make slx SSLTYPE=none

   2. If you will be using SSL to connect to the IMAP server:

           $ make slx EXTRACFLAGS="-I/usr/include/openssl"

   Additionally, you may wish to build on a 64-bit machine, in which case
you need to add -fPIC to EXTRACFLAGS. So, building on a 64-bit machine
with SSL support would look something like:
    $ make slx EXTRACFLAGS="-fPIC -I/usr/include/openssl"

   Or without SSL support:
    $ make slx SSLTYPE=none EXTRACFLAGS=-fPIC

   Once this completes you can proceed with the Asterisk build; there is no
need to run 'make install'.
Compiling Asterisk
Configure with ./configure --with-imap=/usr/src/imap or wherever you built
the UWashington IMAP Toolkit. This directory will be searched for a source
installation. If no source installation is found there, then a package installa-
tion of the IMAP c-client will be searched for in this directory. If one is not
found, then configure will fail.
   A second configure option is to not specify a directory (i.e. ./configure
--with-imap). This will assume that you have the imap-2007e source installed
in the ../imap directory relative to the Asterisk source. If you do not have

                                           163


this source, then configure will default to the "system" option defined in the
next paragraph
    A third option is ./configure --with-imap=system. This will assume that
you have installed a dynamically linked version of the c-client library (most
likely via a package provided by your distro). This will attempt to link agains
-lc-client and will search for c-client headers in your include path starting with
the imap directory, and upon failure, in the c-client directory.
    When you run 'make menuselect', choose 'Voicemail Build Options' and
the IMAP STORAGE option should be available for selection.
    After selecting the IMAP STORAGE option, use the 'x' key to exit
menuselect and save your changes, and the build/install Asterisk normally.

11.2.2 Modify voicemail.conf
The following directives have been added to voicemail.conf:
  imapserver=<name or IP address of IMAP mail server>
  imapport=<IMAP port, defaults to 143>
  imapflags=<IMAP flags, "novalidate-cert" for example>
  imapfolder=<IMAP folder to store messages to>
  imapgreetings=<yes or no>
  greetingsfolder=<IMAP folder to store greetings in if imapgreetings is enabled>
  expungeonhangup=<yes or no>
  authuser=<username>
  authpassword=<password>
  opentimeout=<TCP open timeout in seconds>
  closetimeout=<TCP close timeout in seconds>
  readtimeout=<TCP read timeout in seconds>
  writetimeout=<TCP write timeout in seconds>
    The "imapfolder" can be used to specify an alternative folder on your
IMAP server to store voicemails in. If not specified, the default folder 'IN-
BOX' will be used.
    The "imapgreetings" parameter can be enabled in order to store voicemail
greetings on the IMAP server. If disabled, then they will be stored on the
local file system as normal.
    The "greetingsfolder" can be set to store greetings on the IMAP server
when "imapgreetings" is enabled in an alternative folder than that set by
"imapfolder" or the default folder for voicemails.
    The "expungeonhangup" flag is used to determine if the voicemail system
should expunge all messages marked for deletion when the user hangs up the
phone.
    Each mailbox definition should also have imapuser=<imap username>.
For example:

                                             164


  4123=>4123,James Rothenberger,jar@onebiztone.com,,attach=yes|imapuser=jar
    The directives "authuser" and "authpassword" are not needed when using
Kerberos. They are defined to allow Asterisk to authenticate as a single user
that has access to all mailboxes as an alternative to Kerberos.

11.2.3 IMAP Folders
Besides INBOX, users should create "Old", "Work", "Family" and "Friends"
IMAP folders at the same level of hierarchy as the INBOX. These will be
used as alternate folders for storing voicemail messages to mimic the behavior
of the current (file-based) voicemail system.
    Please note that it is not recommended to store your voicemails in the
top level folder where your users will keep their emails, especially if there
are a large number of emails. A large number of emails in the same folder(s)
that you're storing your voicemails could cause a large delay as Asterisk must
parse through all the emails. For example a mailbox with 100 emails in it
could take up to 60 seconds to receive a response.

11.2.4 Separate vs. Shared Email Accounts
As administrator you will have to decide if you want to send the voicemail
messages to a separate IMAP account or use each user's existing IMAP
mailbox for voicemail storage. The IMAP storage mechanism will work either
way.
    By implementing a single IMAP mailbox, the user will see voicemail mes-
sages appear in the same INBOX as other messages. The disadvantage of
this method is that if the IMAP server does NOT support UIDPLUS, Aster-
isk voicemail will expunge ALL messages marked for deletion when the user
exits the voicemail system, not just the VOICEMAIL messages marked for
deletion.
    By implementing separate IMAPmailboxes for voicemail and email, voice-
mail expunges will not remove regular email flagged for deletion.

11.2.5 IMAP Server Implementations
There are various IMAP server implementations, each supports a potentially
different set of features.

                                            165


UW IMAP-2005 or earlier
UIDPLUSis currently NOTsupported on these versions of UW-IMAP. Please
note that without UID EXPUNGE, Asterisk voicemail will expunge ALL
messages marked for deletion when a user exits the voicemail system (hangs
up the phone).
    This version is *not* recommended for Asterisk.

UW IMAP-2006
This version supports UIDPLUS, which allows UID EXPUNGE capabilities.
This feature allow the system to expunge ONLY pertinent messages, in-
stead of the default behavior, which is to expunge ALL messages marked for
deletion when EXPUNGE is called. The IMAP storage mechanism is this
version of Asterisk will check if the UID EXPUNGE feature is supported by
the server, and use it if possible.
    This version is *not* recommended for Asterisk.
UW IMAP-2007

This is the currently recommended version for use with Asterisk.
Cyrus IMAP
Cyrus IMAP server v2.3.3 has been tested using a hierarchy delimiter of '/'.

11.2.6 Quota Support
If the IMAP server supports quotas, Asterisk will check the quota when
accessing voicemail. Currently only a warning is given to the user that their
quota is exceeded.

11.2.7 Application Notes
Since the primary storage mechanism is IMAP, all message information that
was previously stored in an associated text file, AND the recording itself, is
now stored in a single email message. This means that the .gsm recording
will ALWAYS be attached to the message (along with the user's preference of


                                           166


recording format if different - ie. .WAV). The voicemail message information
is stored in the email message headers. These headers include:
X-Asterisk-VM-Message-Num
X-Asterisk-VM-Server-Name
X-Asterisk-VM-Context
X-Asterisk-VM-Extension
X-Asterisk-VM-Priority
X-Asterisk-VM-Caller-channel
X-Asterisk-VM-Caller-ID-Num
X-Asterisk-VM-Caller-ID-Name
X-Asterisk-VM-Duration
X-Asterisk-VM-Category
X-Asterisk-VM-Orig-date
X-Asterisk-VM-Orig-time



















                                        167




Chapter 12

SMS


12.1 Introduction

The SMS module for Asterisk was developed by Adrian Kennard, and is
an implementation of the ETSI specification for landline SMS, ETSI ES
201 912, which is available from www.etsi.org. Landline SMS is starting
to be available in various parts of Europe, and is available from BT in the
UK. However, Asterisk would allow gateways to be created in other locations
such as the US, and use of SMS capable phones such as the Magic Messenger.
SMS works using analogue or ISDN lines.

12.2 Background
Short Message Service (SMS), or texting is very popular between mobile
phones. A message can be sent between two phones, and normally contains
160 characters. There are ways in which various types of data can be encoded
in a text message such as ring tones, and small graphic, etc. Text messaging
is being used for voting and competitions, and also SPAM...
    Sending a message involves the mobile phone contacting a message centre
(SMSC) and passing the message to it. The message centre then contacts
the destination mobile to deliver the message. The SMSC is responsible
for storing the message and trying to send it until the destination mobile is
available, or a timeout.
    Landline SMS works in basically the same way. You would normally have

                                        168


a suitable text capable landline phone, or a separate texting box such as a
Magic Messenger on your phone line. This sends a message to a message
centre your telco provides by making a normal call and sending the data
using 1200 Baud FSK signaling according to the ETSI spec. To receive a
message the message centre calls the line with a specific calling number, and
the text capable phone answers the call and receives the data using 1200
Baud FSK signaling. This works particularly well in the UK as the calling
line identity is sent before the first ring, so no phones in the house would
ring when a message arrives.

12.3 Typical use with Asterisk
Sending messages from an Asterisk box can be used for a variety of reasons,
including notification from any monitoring systems, email subject lines, etc.
   Receiving messages to an Asterisk box is typically used just to email the
messages to someone appropriate - we email and texts that are received to
our direct numbers to the appropriate person. Received messages could also
be used to control applications, manage competitions, votes, post items to
IRC, anything.
   Using a terminal such as a magic messenger, an Asterisk box could ask
as a message centre sending messages to the terminal, which will beep and
pop up the message (and remember 100 or so messages in its memory).

12.4 Terminology

   * SMS - Short Message Service i.e. text messages
   * SMSC - Short Message Service Centre The system responsible for stor-
      ing and forwarding messages
   * MO - Mobile Originated A message on its way from a mobile or landline
      device to the SMSC
   * MT - Mobile Terminated A message on its way from the SMSC to the
      mobile or landline device

   * RX - Receive A message coming in to the Asterisk box
   * TX - Transmit A message going out of the Asterisk box

                                         169


12.5 Sub address

When sending a message to a landline, you simply send to the landline num-
ber. In the UK, all of the mobile operators (bar one) understand sending
messages to landlines and pass the messages to the BTText system for de-
livery to the landline.
    The specification for landline SMS allows for the possibility of more than
one device on a single landline. These can be configured with Sub addresses
which are a single digit. To send a message to a specific device the message
is sent to the landline number with an extra digit appended to the end. The
telco can define a default sub address (9 in the UK) which is used when the
extra digit is not appended to the end. When the call comes in, part of the
calling line ID is the sub address, so that only one device on the line answers
the call and receives the message.
    Sub addresses also work for outgoing messages. Part of the number called
by the device to send a message is its sub address. Sending from the default
sub address (9 in the UK) means the message is delivered with the sender
being the normal landline number. Sending from any other sub address
makes the sender the landline number with an extra digit on the end.
    Using Asterisk, you can make use of the sub addresses for sending and
receiving messages. Using DDI (DID, i.e. multiple numbers on the line on
ISDN) you can also make use of many different numbers for SMS.

12.6 extensions.conf

The following contexts are recommended.
   ; Mobile Terminated, RX. This is used when an incoming call from the SMS arrive
   s, with the queue (called number and sub address) in ${EXTEN}
   ; Running an app after receipt of the text allows the app to find all messages
   in the queue and handle them, e.g. email them.
   ; The app may be something like smsq --process=somecommand --queue=${EXTEN}
   to run a command for each received message
   ; See below for usage
   [smsmtrx]
   exten = _X.,1, SMS(${EXTEN},a)
   exten = _X.,2,System("someapptohandleincomingsms ${EXTEN}")
   exten = _X.,3,Hangup
   ; Mobile originated, RX. This is receiving a message from a device, e.g.
   ; a Magic Messenger on a sip extension
   ; Running an app after receipt of the text allows the app to find all messages
   ; in the queue and handle then, e.g. sending them to the public SMSC
   ; The app may be something like smsq --process=somecommand --queue=${EXTEN}

                                            170


  ; to run a command for each received message
  ; See below for example usage
  [smsmorx]
  exten = _X.,1, SMS(${EXTEN},sa)
  exten = _X.,2,System("someapptohandlelocalsms ${EXTEN}")
  exten = _X.,3,Hangup
    smsmtrx is normally accessed by an incoming call from the SMSC. In
the UK this call is from a CLI of 080058752X0 where X is the sub address.
As such a typical usage in the extensions.conf at the point of handling an
incoming call is:
  exten = _X./8005875290,1,Goto(smsmtrx,${EXTEN},1)
  exten = _X./_80058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERID(num):8:1},1)

    Alternatively, if you have the correct national prefix on incoming CLI,
e.g. using dahdi hfc, you might use:
  exten = _X./08005875290,1,Goto(smsmtrx,${EXTEN},1)
  exten = _X./_080058752[0-8]0,1,Goto(smsmtrx,${EXTEN}-${CALLERID(num):9:1},1)
    smsmorx is normally accessed by a call from a local sip device connected
to a Magic Messenger. It could however by that you are operating Asterisk
as a message centre for calls from outside. Either way, you look at the called
number and goto smsmorx. In the UK, the SMSC number that would be
dialed is 1709400X where X is the caller sub address. As such typical usage
in extension.config at the point of handling a call from a sip phone is:

  exten = 17094009,1,Goto(smsmorx,${CALLERID(num)},1)
  exten = _1709400[0-8],1,Goto(smsmorx,${CALLERID(num)}-{EXTEN:7:1},1)

12.7 Using smsq
smsq is a simple helper application designed to make it easy to send messages
from a command line. it is intended to run on the Asterisk box and have
direct access to the queue directories for SMS and for Asterisk.
    In its simplest form you can send an SMS by a command such as smsq
0123456789 This is a test to 0123456789 This would create a queue file
for a mobile originated TX message in queue 0 to send the text "This is
a test to 0123456789" to 0123456789. It would then place a file in the
/var/spool/asterisk/outgoing directory to initiate a call to 17094009 (the
default message centre in smsq) attached to application SMS with argument
of the queue name (0).

                                             171


    Normally smsq will queue a message ready to send, and will then create
a file in the Asterisk outgoing directory causing Asterisk to actually connect
to the message centre or device and actually send the pending message(s).
    Using --process, smsq can however be used on received queues to run
a command for each file (matching the queue if specified) with various envi-
ronment variables set based on the message (see below); smsq options:
    --help
    Show help text
    --usage
    Show usage
    --queue
    -q
    Specify a specific queue
    In no specified, messages are queued under queue "0"
    --da
    -d
    Specify destination address
    --oa
    -o
    Specify originating address
    This also implies that we are generating a mobile terminated message
    --ud
    -m
    Specify the actual message
    --ud-file
    -f
    Specify a file to be read for the context of the message
    A blank filename (e.g. --ud-file= on its own) means read stdin. Very
    useful when using via ssh where command line parsing could mess up the
    message.
    --mt
    -t
    Mobile terminated message to be generated
    --mo
    Mobile originated message to be generated
    Default
    --tx

                                        172


Transmit message
Default
--rx
-r
Generate a message in the receive queue
--UTF-8
Treat the file as UTF-8 encoded (default)
--UCS-1
Treat the file as raw 8 bit UCS-1 data, not UTF-8 encoded
--UCS-2
Treat the file as raw 16 bit bigendian USC-2 data
--process
Specific a command to process for each file in the queue
Implies --rx and --mt if not otherwise specified.
Sets environment variables for every possible variable, and also ud,
ud8 (USC-1 hex), and ud16 (USC-2 hex) for each call. Removes files.
--motx-channel
Specify the channel for motx calls
May contain X to use sub address based on queue name or may be full
number
Default is Local/1709400X
--motx-callerid
Specify the caller ID for motx calls
The default is the queue name without -X suffix
--motx-wait
Wait time for motx call
Default 10
--motx-delay
Retry time for motx call
Default 1
--motx-retries
Retries for motx call
Default 10
--mttx-channel
Specify the channel for mttx calls
Default is Local/ and the queue name without -X suffix
--mtttx-callerid
Specify the callerid for mttx calls

                           173


May include X to use sub address based on queue name or may be full
number
Default is 080058752X0
--mttx-wait
Wait time for mttx call
Default 10
--mttx-delay
Retry time for mttx call
Default 30
--mttx-retries
Retries for mttx call
Default 100
--default-sub-address
The default sub address assumed (e.g. for X in CLI and dialled numbers
as above) when none added (-X) to queue
Default 9
--no-dial
-x
Create queue, but do not dial to send message
--no-wait
Do not wait if a call appears to be in progress
This could have a small window where a message is queued but not
sent, so regular calls to smsq should be done to pick up any missed
messages
--concurrent
How many concurrent calls to allow (per queue), default 1
--mr
-n
Message reference
--pid
-p
Protocol ID
--dcs
Data coding scheme
--udh
Specific hex string of user data header specified (not including the
initial length byte)
May be a blank string to indicate header is included in the user data

                           174


   already but user data header indication to be set.
   --srr
   Status report requested
   --rp
   Return path requested
   --vp
   Specify validity period (seconds)
   --scts
   Specify timestamp (YYYY-MM-DDTHH:MM:SS)
   --spool-dir
   Spool dir (in which sms and outgoing are found)
   Default /var/spool/asterisk
   Other arguments starting '-' or '--' are invalid and will cause an error.
Any trailing arguments are processed as follows:-

   * If the message is mobile originating and no destination address has
      been specified, then the first argument is assumed to be a destination
      address
   * If the message is mobile terminating and no destination address has
      been specified, then the first argument is assumed to be the queue
      name
   * If there is no user data, or user data file specified, then any following
      arguments are assumed to be the message, which are concatenated.

   * If no user data is specified, then no message is sent. However, unless
      --no-dial is specified, smsq checks for pending messages and generates
      an outgoing anyway
   Note that when smsq attempts to make a file in /var/spool/asterisk/
outgoing, it checks if there is already a call queued for that queue. It will
try several filenames, up to the --concurrent setting. If these files exist,
then this means Asterisk is already queued to send all messages for that
queue, and so Asterisk should pick up the message just queued. However,
this alone could create a race condition, so if the files exist then smsq will
wait up to 3 seconds to confirm it still exists or if the queued messages have
been sent already. The --no-wait turns off this behaviour. Basically, this

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means that if you have a lot of messages to send all at once, Asterisk will not
make unlimited concurrent calls to the same message centre or device for the
same queue. This is because it is generally more efficient to make one call
and send all of the messages one after the other.
    smsq can be used with no arguments, or with a queue name only, and it
will check for any pending messages and cause an outgoing if there are any.
It only sets up one outgoing call at a time based on the first queued message
it finds. A outgoing call will normally send all queued messages for that
queue. One way to use smsq would be to run with no queue name (so any
queue) every minute or every few seconds to send pending message. This is
not normally necessary unless --no-dial is selected. Note that smsq does
only check motx or mttx depending on the options selected, so it would need
to be called twice as a general check.
    UTF-8 is used to parse command line arguments for user data, and is
the default when reading a file. If an invalid UTF-8 sequence is found, it
is treated as UCS-1 data (i.e, as is). The --process option causes smsq to
scan the specified queue (default is mtrx) for messages (matching the queue
specified, or any if queue not specified) and run a command and delete the
file. The command is run with a number of environment variables set as
follows. Note that these are unset if not needed and not just taken from the
calling environment. This allows simple processing of incoming messages
    $queue
    Set if a queue specified
    $?srr
    srr is set (to blank) if srr defined and has value 1.
    $?rp
    rp is set (to blank) if rp defined and has value 1.
    $ud
    User data, UTF-8 encoding, including any control characters, but with
    nulls stripped out
    Useful for the content of emails, for example, as it includes any
    newlines, etc.
    $ude
    User data, escaped UTF-8, including all characters, but control
    characters \n, \r, \t, \f, \xxx and \ is escaped as \\
    Useful guaranteed one line printable text, so useful in Subject lines
    of emails, etc

                                            176


    $ud8
    Hex UCS-1 coding of user data (2 hex digits per character)
    Present only if all user data is in range U+0000 to U+00FF
    $ud16
    Hex UCS-2 coding of user data (4 hex digits per character)
    other
    Other fields set using their field name, e.g. mr, pid, dcs, etc. udh
    is a hex byte string

12.8 File formats
By default all queues are held in a director /var/spool/asterisk/sms.
Within this directory are sub directories mtrx, mttx, morx, motx which hold
the received messages and the messages ready to send. Also, /var/log/
asterisk/sms is a log file of all messages handled.
   The file name in each queue directory starts with the queue parameter to
SMS which is normally the CLI used for an outgoing message or the called
number on an incoming message, and may have -X (X being sub address)
appended. If no queue ID is known, then 0 is used by smsq by default. After
this is a dot, and then any text. Files are scanned for matching queue ID and
a dot at the start. This means temporary files being created can be given a
different name not starting with a queue (we recommend a . on the start of
the file name for temp files). Files in these queues are in the form of a simple
text file where each line starts with a keyword and an = and then data. udh
and ud have options for hex encoding, see below.
   UTF-8. The user data (ud) field is treated as being UTF-8 encoded unless
the DCS is specified indicating 8 bit format. If 8 bit format is specified then
the user data is sent as is. The keywords are as follows:

    oa Originating address
    The phone number from which the message came
    Present on mobile terminated messages and is the CLI for morx messages
    da
    Destination Address
    The phone number to which the message is sent
    Present on mobile originated messages
    scts
    The service centre time stamp
                                              177


    Format YYYY-MM-DDTHH:MM:SS
    Present on mobile terminated messages
    pid
    One byte decimal protocol ID
    See GSM specs for more details
    Normally 0 or absent
    dcs
    One byte decimal data coding scheme
    If omitted, a sensible default is used (see below)
    See GSM specs for more details
    mr
    One byte decimal message reference
    Present on mobile originated messages, added by default if absent
    srr
    0 or 1 for status report request
    Does not work in UK yet, not implemented in app_sms yet
    rp
    0 or 1 return path
    See GSM specs for details
    vp
    Validity period in seconds
    Does not work in UK yet
    udh
    Hex string of user data header prepended to the SMS contents,
    excluding initial length byte.
    Consistent with ud, this is specified as udh# rather than udh=
    If blank, this means that the udhi flag will be set but any user data
    header must be in the ud field
    ud
    User data, may be text, or hex, see below
    udh is specified as as udh# followed by hex (2 hex digits per byte). If
present, then the user data header indicator bit is set, and the length plus
the user data header is added to the start of the user data, with padding
if necessary (to septet boundary in 7 bit format). User data can hold an
USC character codes U+0000 to U+FFFF. Any other characters are coded
as U+FEFF
    ud can be specified as ud= followed by UTF-8 encoded text if it contains

                                       178


no control characters, i.e. only (U+0020 to U+FFFF). Any invalid UTF-8
sequences are treated as is (U+0080-U+00FF).
   ud can also be specified as ud# followed by hex (2 hex digits per byte)
containing characters U+0000 to U+00FF only.
   ud can also be specified as ud## followed by hex (4 hex digits per byte)
containing UCS-2 characters.
   When written by app sms (e.g. incoming messages), the file is written
with ud= if it can be (no control characters). If it cannot, the a comment
line ;ud= is used to show the user data for human readability and ud# or
ud## is used.

12.9 Delivery reports
The SMS specification allows for delivery reports. These are requested using
the srr bit. However, as these do not work in the UK yet they are not fully
implemented in this application. If anyone has a telco that does implement
these, please let me know. BT in the UK have a non standard way to do this
by starting the message with *0#, and so this application may have a UK
specific bodge in the near future to handle these.
   The main changes that are proposed for delivery report handling are :

   * New queues for sent messages, one file for each destination address and
      message reference.
   * New field in message format, user reference, allowing applications to
      tie up their original message with a report.
   * Handling of the delivery confirmation/rejection and connecting to the
      outgoing message - the received message file would then have fields for
      the original outgoing message and user reference allowing applications
      to handle confirmations better.






                                           179




Chapter 13


Queues


13.1 Introduction

Pardon, but the dialplan in this tutorial will be expressed in AEL, the new
Asterisk Extension Language. If you are not used to its syntax, we hope you
will find it to some degree intuitive. If not, there are documents explaining
its syntax and constructs.

13.2 Configuring Call Queues

13.2.1 queues.conf
First of all, set up call queues in queue.conf
    Here is an example:
     =========== queues.conf ===========
     | ; Cool Digium Queues |
     | [general] |
     | persistentmembers = yes |
     | |
     | ; General sales queue |
     | [sales-general] |
     | music=default |
     | context=sales |
     | strategy=ringall |
     | joinempty=strict |
     | leavewhenempty=strict |
     | |
     | ; Customer service queue |
     | [customerservice] |

                                           180


     | music=default |
     | context=customerservice |
     | strategy=ringall |
     | joinempty=strict |
     | leavewhenempty=strict |
     | |
     | ; Support dispatch queue |
     | [dispatch] |
     | music=default |
     | context=dispatch |
     | strategy=ringall |
     | joinempty=strict |
     | leavewhenempty=strict |
     ===================================
    In the above, we have defined 3 separate calling queues: sales-general,
customerservice, and dispatch.
    Please note that the sales-general queue specifies a context of "sales",
and that customerservice specifies the context of "customerservice", and the
dispatch queue specifies the context "dispatch". These three contexts must
be defined somewhere in your dialplan. We will show them after the main
menu below.
    In the [general] section, specifying the persistentmembers=yes, will cause
the agent lists to be stored in astdb, and recalled on startup.
    The strategy=ringall will cause all agents to be dialed together, the first
to answer is then assigned the incoming call.
    "joinempty" set to "strict" will keep incoming callers from being placed
in queues where there are no agents to take calls. The Queue() application
will return, and the dial plan can determine what to do next.
    If there are calls queued, and the last agent logs out, the remaining in-
coming callers will immediately be removed from the queue, and the Queue()
call will return, IF the "leavewhenempty" is set to "strict".

13.2.2 Routing incoming Calls to Queues

Then in extensions.ael, you can do these things:
The Main Menu

At Digium, incoming callers are sent to the "mainmenu" context, where they
are greeted, and directed to the numbers they choose...
  context mainmenu {

                                           181


  includes {
  digium;
  queues-loginout;
  }
          0 => goto dispatch,s,1;
          2 => goto sales,s,1;
          3 => goto customerservice,s,1;
          4 => goto dispatch,s,1;

          s => {
                  Ringing();
                  Wait(1);
                  Set(attempts=0);
                  Answer();
                  Wait(1);
                  Background(digium/ThankYouForCallingDigium);
                  Background(digium/YourOpenSourceTelecommunicationsSupplier);
                  WaitExten(0.3);
          repeat:
                  Set(attempts=$[${attempts} + 1]);
                  Background(digium/IfYouKnowYourPartysExtensionYouMayDialItAtAnyTime);
                  WaitExten(0.1);
                  Background(digium/Otherwise);
                  WaitExten(0.1);
                  Background(digium/ForSalesPleasePress2);
                  WaitExten(0.2);
                  Background(digium/ForCustomerServicePleasePress3);
                  WaitExten(0.2);
                  Background(digium/ForAllOtherDepartmentsPleasePress4);
                  WaitExten(0.2);
                  Background(digium/ToSpeakWithAnOperatorPleasePress0AtAnyTime);
                  if( ${attempts} < 2 ) {
                          WaitExten(0.3);
                          Background(digium/ToHearTheseOptionsRepeatedPleaseHold);
                  }
                  WaitExten(5);
                  if( ${attempts} < 2 ) goto repeat;
                  Background(digium/YouHaveMadeNoSelection);
                  Background(digium/ThisCallWillBeEnded);
                  Background(goodbye);
                  Hangup();
          }
  }

The Contexts referenced from the queues.conf file
  context sales {

          0 => goto dispatch,s,1;
          8 => Voicemail(${SALESVM});
          s => {
                  Ringing();
                  Wait(2);


                                           182


                  Background(digium/ThankYouForContactingTheDigiumSalesDepartment);
                  WaitExten(0.3);
                  Background(digium/PleaseHoldAndYourCallWillBeAnsweredByOurNextAvailableSalesRepresentative);
                  WaitExten(0.3);
                  Background(digium/AtAnyTimeYouMayPress0ToSpeakWithAnOperatorOr8ToLeaveAMessage);
                  Set(CALLERID(name)=Sales);
                  Queue(sales-general,t);
                  Set(CALLERID(name)=EmptySalQ);
                  goto dispatch,s,1;
                  Playback(goodbye);
                  Hangup();
          }
  }

    Please note that there is only one attempt to queue a call in the sales
queue. All sales agents that are logged in will be rung.
  context customerservice {

          0 => {
                  SetCIDName(CSVTrans);
                  goto dispatch|s|1;
          }
          8 => Voicemail(${CUSTSERVVM});
          s => {
                  Ringing();
                  Wait(2);
                  Background(digium/ThankYouForCallingDigiumCustomerService);
                  WaitExten(0.3);
          notracking:
                  Background(digium/PleaseWaitForTheNextAvailableCustomerServiceRepresentative);
                  WaitExten(0.3);
                  Background(digium/AtAnyTimeYouMayPress0ToSpeakWithAnOperatorOr8ToLeaveAMessage);
                  Set(CALLERID(name)=Cust Svc);
                  Set(QUEUE_MAX_PENALTY=10);
                  Queue(customerservice,t);
                  Set(QUEUE_MAX_PENALTY=0);
                  Queue(customerservice,t);
                  Set(CALLERID(name)=EmptyCSVQ);
                  goto dispatch,s,1;
                  Background(digium/NoCustomerServiceRepresentativesAreAvailableAtThisTime);
                  Background(digium/PleaseLeaveAMessageInTheCustomerServiceVoiceMailBox);
                  Voicemail(${CUSTSERVVM});
                  Playback(goodbye);
                  Hangup();
          }
  }

    Note that calls coming into customerservice will first be try to queue
calls to those agents with a QUEUE MAX PENALTY of 10, and if none are
available, then all agents are rung.



                                           183


  context dispatch
  {
          s => {
                  Ringing();
                  Wait(2);
                  Background(digium/ThankYouForCallingDigium);
                  WaitExten(0.3);
                  Background(digium/YourCallWillBeAnsweredByOurNextAvailableOperator);
                  Background(digium/PleaseHold);
                  Set(QUEUE_MAX_PENALTY=10);
                  Queue(dispatch|t);
                  Set(QUEUE_MAX_PENALTY=20);
                  Queue(dispatch|t);
                  Set(QUEUE_MAX_PENALTY=0);
                  Queue(dispatch|t);
                  Background(digium/NoOneIsAvailableToTakeYourCall);
                  Background(digium/PleaseLeaveAMessageInOurGeneralVoiceMailBox);
                  Voicemail(${DISPATCHVM});
                  Playback(goodbye);
                  Hangup();
          }
  }
   And in the dispatch context, first agents of priority 10 are tried, then 20,
and if none are available, all agents are tried.
   Notice that a common pattern is followed in each of the three queue
contexts:
   First, you set QUEUE MAX PENALTYto a value, then you call Queue(<queue-
name>,option,...) (see the Queue application documetation for details)
   In the above, note that the "t" option is specified, and this allows the
agent picking up the incoming call the luxury of transferring the call to other
parties.
   The purpose of specifying the QUEUE MAX PENALTY is to develop
a set of priorities amongst agents. By the above usage, agents with lower
number priorities will be given the calls first, and then, if no-one picks up
the call, the QUEUE MAX PENALTY will be incremented, and the queue
tried again. Hopefully, along the line, someone will pick up the call, and the
Queue application will end with a hangup.
   The final attempt to queue in most of our examples sets the QUEUE -
MAX PENALTY to zero, which means to try all available agents.

13.2.3 Assigning agents to Queues

In this example dialplan, we want to be able to add and remove agents to
handle incoming calls, as they feel they are available. As they log in, they
                                           184


are added to the queue's agent list, and as they log out, they are removed.
If no agents are available, the queue command will terminate, and it is the
duty of the dialplan to do something appropriate, be it sending the incoming
caller to voicemail, or trying the queue again with a higher QUEUE MAX -
PENALTY.
    Because a single agent can make themselves available to more than one
queue, the process of joining multiple queues can be handled automatically
by the dialplan.

Agents Log In and Out
  context queues-loginout
  {
          6092 => {
                           Answer();
                           Read(AGENT_NUMBER,agent-enternum);
                           VMAuthenticate(${AGENT_NUMBER}@default,s);
                           Set(queue-announce-success=1);
                           goto queues-manip,I${AGENT_NUMBER},1;
                   }
          6093 => {
                           Answer();
                           Read(AGENT_NUMBER,agent-enternum);
                           Set(queue-announce-success=1);
                           goto queues-manip,O${AGENT_NUMBER},1;
                   }
  }

    In the above contexts, the agents dial 6092 to log into their queues, and
they dial 6093 to log out of their queues. The agent is prompted for their
agent number, and if they are logging in, their passcode, and then they
are transferred to the proper extension in the queues-manip context. The
queues-manip context does all the actual work:
  context queues-manip {
          // Raquel Squelch
          _[IO]6121 => {
                   &queue-addremove(dispatch,10,${EXTEN});
                   &queue-success(${EXTEN});
          }
          // Brittanica Spears
          _[IO]6165 => {
                   &queue-addremove(dispatch,20,${EXTEN});
                   &queue-success(${EXTEN});
          }


                                            185


          // Rock Hudson
          _[IO]6170 => {
                  &queue-addremove(sales-general,10,${EXTEN});
                  &queue-addremove(customerservice,20,${EXTEN});
                  &queue-addremove(dispatch,30,${EXTEN});
                  &queue-success(${EXTEN});
          }
          // Saline Dye-on
          _[IO]6070 => {
                  &queue-addremove(sales-general,20,${EXTEN});
                  &queue-addremove(customerservice,30,${EXTEN});
                  &queue-addremove(dispatch,30,${EXTEN});
                  &queue-success(${EXTEN});
          }
  }

    In the above extensions, note that the queue-addremove macro is used
to actually add or remove the agent from the applicable queue, with the
applicable priority level. Note that agents with a priority level of 10 will be
called before agents with levels of 20 or 30.
    In the above example, Raquel will be dialed first in the dispatch queue, if
she has logged in. If she is not, then the second call of Queue() with priority
of 20 will dial Brittanica if she is present, otherwise the third call of Queue()
with MAX PENALTY of 0 will dial Rock and Saline simultaneously.
    Also note that Rock will be among the first to be called in the sales-
general queue, and among the last in the dispatch queue. As you can see
in main menu, the callerID is set in the main menu so they can tell which
queue incoming calls are coming from.
    The call to queue-success() gives some feedback to the agent as they log
in and out, that the process has completed.
  macro queue-success(exten)
  {
          if( ${queue-announce-success} > 0 )
          {
                  switch(${exten:0:1})
                  {
                  case I:
                          Playback(agent-loginok);
                          Hangup();
                          break;
                  case O:
                          Playback(agent-loggedoff);
                          Hangup();
                          break;
                  }
          }
  }

                                            186


   The queue-addremove macro is defined in this manner:

  macro queue-addremove(queuename,penalty,exten)
  {
          switch(${exten:0:1})
          {
          case I: // Login
                   AddQueueMember(${queuename},Local/${exten:1}@agents,${penalty});
   break;
          case O: // Logout
                   RemoveQueueMember(${queuename},Local/${exten:1}@agents);
   break;
          case P: // Pause
                   PauseQueueMember(${queuename},Local/${exten:1}@agents);
   break;
          case U: // Unpause
                   UnpauseQueueMember(${queuename},Local/${exten:1}@agents);
   break;
          default: // Invalid
                   Playback(invalid);
                   break;
          }
  }
   Basically, it uses the first character of the exten variable, to determine
the proper actions to take. In the above dial plan code, only the cases I or
O are used, which correspond to the Login and Logout actions.


13.2.4 Controlling The Way Queues Call the Agents
Notice in the above, that the commands to manipulate agents in queues have
"@agents" in their arguments. This is a reference to the agents context:

  context agents
  {
  // General sales queue
  8010 =>
  {
  Set(QUEUE_MAX_PENALTY=10);
  Queue(sales-general,t);
  Set(QUEUE_MAX_PENALTY=0);
  Queue(sales-general,t);
  Set(CALLERID(name)=EmptySalQ);
  goto dispatch,s,1;
  }
  // Customer Service queue
  8011 =>
  {
  Set(QUEUE_MAX_PENALTY=10);
  Queue(customerservice,t);
  Set(QUEUE_MAX_PENALTY=0);
  Queue(customerservice,t);

                                           187


  Set(CALLERID(name)=EMptyCSVQ);
  goto dispatch,s,1;
  }
  8013 =>
  {
  Dial(iax2/sweatshop/9456@from-ecstacy);
  Set(CALLERID(name)=EmptySupQ);
  Set(QUEUE_MAX_PENALTY=10);
  Queue(support-dispatch,t);
  Set(QUEUE_MAX_PENALTY=20);
  Queue(support-dispatch,t);
  Set(QUEUE_MAX_PENALTY=0); // means no max
  Queue(support-dispatch,t);
  goto dispatch,s,1;
  }
  6121 => &callagent(${RAQUEL},${EXTEN});
  6165 => &callagent(${SPEARS},${EXTEN});
  6170 => &callagent(${ROCK},${EXTEN});
  6070 => &callagent(${SALINE},${EXTEN});
  }

   In the above, the variables ${RAQUEL}, etc stand for actual devices to
ring that person's phone (like DAHDI/37).
   The 8010, 8011, and 8013 extensions are purely for transferring incoming
callers to queues. For instance, a customer service agent might want to
transfer the caller to talk to sales. The agent only has to transfer to extension
8010, in this case.
   Here is the callagent macro, note that if a person in the queue is called,
but does not answer, then they are automatically removed from the queue.
  macro callagent(device,exten)
  {
  if( ${GROUP_COUNT(${exten}@agents)}=0 )
  {
  Set(OUTBOUND_GROUP=${exten}@agents);
  Dial(${device},300,t);
  switch(${DIALSTATUS})
  {
  case BUSY:
  Busy();
  break;
  case NOANSWER:
  Set(queue-announce-success=0);
  goto queues-manip,O${exten},1;
  default:
  Hangup();
  break;
  }
  }
  else
  {
  Busy();

                                           188


  }
  }
    In the callagent macro above, the ${exten} will be 6121, or 6165, etc,
which is the extension of the agent.
    The use of the GROUP COUNT, and OUTBOUND GROUP follow this
line of thinking. Incoming calls can be queued to ring all agents in the
current priority. If some of those agents are already talking, they would get
bothersome call-waiting tones. To avoid this inconvenience, when an agent
gets a call, the OUTBOUND GROUP assigns that conversation to the group
specified, for instance 6171@agents. The ${GROUP COUNT()} variable on
a subsequent call should return "1" for that group. If GROUP COUNT
returns 1, then the busy() is returned without actually trying to dial the
agent.


13.2.5 Pre Acknowledgement Message
If you would like to have a pre acknowledge message with option to reject
the message you can use the following dialplan Macro as a base with the 'M'
dial argument.
  [macro-screen]
  exten=>s,1,Wait(.25)
  exten=>s,2,Read(ACCEPT,screen-callee-options,1)
  exten=>s,3,Gotoif($[${ACCEPT} = 1] ?50)
  exten=>s,4,Gotoif($[${ACCEPT} = 2] ?30)
  exten=>s,5,Gotoif($[${ACCEPT} = 3] ?40)
  exten=>s,6,Gotoif($[${ACCEPT} = 4] ?30:30)
  exten=>s,30,Set(MACRO_RESULT=CONTINUE)
  exten=>s,40,Read(TEXTEN,custom/screen-exten,)
  exten=>s,41,Gotoif($[${LEN(${TEXTEN})} = 3]?42:45)
  exten=>s,42,Set(MACRO_RESULT=GOTO:from-internal^${TEXTEN}^1)
  exten=>s,45,Gotoif($[${TEXTEN} = 0] ?46:4)
  exten=>s,46,Set(MACRO_RESULT=CONTINUE)
  exten=>s,50,Playback(after-the-tone)
  exten=>s,51,Playback(connected)
  exten=>s,52,Playback(beep)

13.2.6 Caveats

In the above examples, some of the possible error checking has been omitted,
to reduce clutter and make the examples clearer.


                                            189


13.3 Queue Logs
In order to properly manage ACD queues, it is important to be able to keep
track of details of call setups and teardowns in much greater detail than
traditional call detail records provide. In order to support this, extensive
and detailed tracing of every queued call is stored in the queue log, located
(by default) in /var/log/asterisk/queue_log.
    These are the events (and associated information) in the queue log:
    ABANDON(position|origposition|waittime)
    The caller abandoned their position in the queue. The position is the
caller's position in the queue when they hungup, the origposition is the origi-
nal position the caller was when they first entered the queue, and the waittime
is how long the call had been waiting in the queue at the time of disconnect.
    AGENTDUMP
    The agent dumped the caller while listening to the queue announcement.
    AGENTLOGIN(channel)
    The agent logged in. The channel is recorded.
    AGENTCALLBACKLOGIN(exten@context)
    The callback agent logged in. The login extension and context is recorded.
    AGENTLOGOFF(channel|logintime)
    The agent logged off. The channel is recorded, along with the total time
the agent was logged in.
    AGENTCALLBACKLOGOFF(exten@context|logintime|reason)
    The callback agent logged off. The last login extension and context is
recorded, along with the total time the agent was logged in, and the reason
for the logoff if it was not a normal logoff (e.g., Autologoff, Chanunavail)
    COMPLETEAGENT(holdtime|calltime|origposition)
    The caller was connected to an agent, and the call was terminated nor-
mally by the *agent*. The caller's hold time and the length of the call are
both recorded. The caller's original position in the queue is recorded in
origposition.
    COMPLETECALLER(holdtime|calltime|origposition)
    The caller was connected to an agent, and the call was terminated nor-
mally by the *caller*. The caller's hold time and the length of the call are
both recorded. The caller's original position in the queue is recorded in
origposition.
    CONFIGRELOAD
    The configuration has been reloaded (e.g. with asterisk -rx reload)

                                           190


    CONNECT(holdtime|bridgedchanneluniqueid|ringtime)
    The caller was connected to an agent. Hold time represents the amount
of time the caller was on hold. The bridged channel unique ID contains the
unique ID of the queue member channel that is taking the call. This is useful
when trying to link recording filenames to a particular call in the queue.
Ringtime represents the time the queue members phone was ringing prior to
being answered.
    ENTERQUEUE(url|callerid)
    A call has entered the queue. URL (if specified) and Caller*ID are placed
in the log.
    EXITEMPTY(position|origposition|waittime)
    The caller was exited from the queue forcefully because the queue had
no reachable members and it's configured to do that to callers when there
are no reachable members. The position is the caller's position in the queue
when they hungup, the origposition is the original position the caller was
when they first entered the queue, and the waittime is how long the call had
been waiting in the queue at the time of disconnect.
    EXITWITHKEY(key|position|origposition|waittime)
    The caller elected to use a menu key to exit the queue. The key and the
caller's position in the queue are recorded. The caller's entry position and
amoutn of time waited is also recorded.
    EXITWITHTIMEOUT(position|origposition|waittime)
    The caller was on hold too long and the timeout expired. The position
in the queue when the timeout occurred, the entry position, and the amount
of time waited are logged.
    QUEUESTART
    The queueing system has been started for the first time this session.
    RINGNOANSWER(ringtime)
    After trying for ringtime ms to connect to the available queue member,
the attempt ended without the member picking up the call. Bad queue
member!
    SYSCOMPAT
    A call was answered by an agent, but the call was dropped because the
channels were not compatible.
    TRANSFER(extension|context|holdtime|calltime|origposition)
    Caller was transferred to a different extension. Context and extension are
recorded. The caller's hold time and the length of the call are both recorded,
as is the caller's entry position at the time of the transfer. PLEASE remember

                                            191


that transfers performed by SIP UA's by way of a reinvite may not always
be caught by Asterisk and trigger off this event. The only way to be 100%
sure that you will get this event when a transfer is performed by a queue
member is to use the built-in transfer functionality of Asterisk.


























                                       192




Chapter 14

Phone Provisioning


14.1 Introduction

Asterisk includes basic phone provisioning support through the res phone-
prov module. The current implementation is based on a templating sys-
tem using Asterisk dialplan function and variable substitution and obtains
information to substitute into those templates from phoneprov.conf and
users.conf. A profile and set of templates is provided for provisioning Poly-
com phones. Note that res phoneprov is currently limited to provisioning a
single user per device.

14.2 Configuration of phoneprov.conf
The configuration file, phoneprov.conf, is used to set up the built-in vari-
ables SEVER and SERVER PORT, to define a default phone profile to use,
and to define different phone profiles available for provisioning.

14.2.1 The [general] section
Below is a sample of the general section of phoneprov.conf:
  [general]
  ;serveriface=eth0
  ;serveraddr=192.168.1.1
  ;serverport=5060
  default_profile=polycom

                                          193


    By default, res phoneprov will set the SERVER variable to the IP address
on the server that the requesting phone uses to contact the asterisk HTTP
server. The SERVER PORT variable will default to the bindport setting
in sip.conf.
    Should the defaults be insufficient, there are two choices for overriding
the default setting of the SERVER variable. If the IP address of the server is
known, or the hostname resolvable by the phones, the appropriate server-
addr value should be set. Alternatively, the network interface that the server
listens on can be set by specifying a serveriface and SERVER will be set to
the IP address of that interface. Only one of these options should be set.
    The default SERVER PORT variable can be overridden by setting the
serverport. If bindport is not set in sip.conf and serverport is not spec-
ified, it is set to a default value of 5060.
    Any user set for auto-provisioning in users.conf without a specified profile
will be assumed to belong to the profile set with default profile.

14.2.2 Creating phone profiles
A phone profile is basically a list of files that a particular group of phones
needs to function. For most phone types there are files that are identical
for all phones (firmware, for instance) as well as a configuration file that is
specific to individual phones. res phoneprov breaks these two groups of files
into static files and dynamic files, respectively. A sample profile:

  [polycom]
  staticdir => configs/
  mime_type => text/xml
  setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg
  static_file => bootrom.ld,application/octet-stream
  static_file => bootrom.ver,plain/text
  static_file => sip.ld,application/octet-stream
  static_file => sip.ver,plain/text
  static_file => sip.cfg
  static_file => custom.cfg
  ${TOLOWER(${MAC})}.cfg => 000000000000.cfg
  ${TOLOWER(${MAC})}-phone.cfg => 000000000000-phone.cfg
  config/${TOLOWER(${MAC})} => polycom.xml
  ${TOLOWER(${MAC})}-directory.xml => 000000000000-directory.xml
    A static file is set by specifying the file name, relative to AST\_DATA\
_DIR/phoneprov. The mime-type of the file can optionally be specified after
a comma. If staticdir is set, all static files will be relative to the subdirectory
of AST DATA DIR/phoneprov specified.

                                              194


    Since phone-specific config files generally have file names based on phone-
specifc data, dynamic filenames in res phoneprov can be defined with As-
terisk dialplan function and variable substitution. In the above example,
${TOLOWER(${MAC})}.cfg => 000000000000.cfg would define a relative
URI to be served that matches the format of MACADDRESS.cfg, all lower
case. A request for that file would then point to the template found at AST -
DATA DIR/phoneprov/000000000000.cfg. The template can be followed by
a comma and mime-type. Notice that the dynamic filename (URI) can con-
tain contain directories. Since these files are dynamically generated, the
config file itself does not reside on the filesystem--only the template. To view
the generated config file, open it in a web browser. If the config file is XML,
Firefox should display it. Some browsers will require viewing the source of
the page requested.
    A default mime-type for the profile can be defined by setting mime-
type. If a custom variable is required for a template, it can be specified with
setvar. Variable substitution on this value is done while building the route
list, so ${USERNAME} would expand to the username of the users.conf user
that registers the dynamic filename.
    NOTE: Any dialplan function that is used for generation of dynamic file
names MUST be loaded before res phoneprov. Add "preload => module-
name.so" to modules.conf for required functions. In the example above,
"preload => func strings.so" would be required.

14.3 Configuration of users.conf
The asterisk-gui sets up extensions, SIP/IAX2 peers, and a host of other
settings. User-specific settings are stored in users.conf. If the asterisk-gui is
not being used, manual entries to users.conf can be made.

14.3.1 The [general] section
There are only two settings in the general section of users.conf that apply
to phone provisioning: localextenlength which maps to template variable
EXTENSION LENGTH and vmexten which maps to the VOICEMAIL -
EXTEN variable.



                                              195


14.3.2 Invdividual Users

To enable auto-provisioning of a phone, the user in users.conf needs to
have:
  ...
  autoprov=yes
  macaddress=deadbeef4dad
  profile=polycom

   The profile is optional if a default profile is set in phoneprov.conf. The
following is a sample users.conf entry, with the template variables commented
next to the settings:
  [6001]
  callwaiting = yes
  context = numberplan-custom-1
  hasagent = no
  hasdirectory = yes
  hasiax = no
  hasmanager = no
  hassip = yes
  hasvoicemail = yes
  host = dynamic
  mailbox = 6001
  threewaycalling = yes
  deletevoicemail = no
  autoprov = yes
  profile = polycom
  directmedia = no
  nat = no
  fullname = User Two ; ${DISPLAY_NAME}
  secret = test ; ${SECRET}
  username = 6001 ; ${USERNAME}
  macaddress = deadbeef4dad ; ${MAC}
  label = 6001 ; ${LABEL}
  cid_number = 6001 ; ${CALLERID}

   The variables above, are the user-specfic variables that can be substituted
into dynamic filenames and config templates.

14.4 Templates

Configuration templates are a generic way to configure phones with text-
based configuration files. Templates can use any loaded dialplan function
and all of the variables created by phoneprov.conf and users.conf. A
short example is the included 000000000000.cfg Polycom template:

                                            196


  <?xml version="1.0" standalone="yes"?>
    <APPLICATION
      APP_FILE_PATH="sip.ld"
      CONFIG_FILES="${IF($[${STAT(e|${CUSTOM_CONFIG})}] ? "custom.cfg,
  ")}config/${TOLOWER(${MAC})}, sip.cfg"
      MISC_FILES="" LOG_FILE_DIRECTORY=""
    />
   This template uses dialplan functions, expressions, and a couple of vari-
ables to generate a config file to instruct the Polycom where to pull other
needed config files. If a phone with MAC address 0xDEADBEEF4DAD re-
quests this config file, and the filename that is stored in variable CUSTOM -
CONFIG does not exist, then the generated output would be:

  <?xml version="1.0" standalone="yes"?>
    <APPLICATION
      APP_FILE_PATH="sip.ld"
      CONFIG_FILES="config/deadbeef4dad, sip.cfg"
      MISC_FILES="" LOG_FILE_DIRECTORY=""
    />
   The Polycom phone would then download both sip.cfg (which would
be registered in phoneprov.conf as a static file) and config/deadbeef4dad
(which would be registered as a dynamic file pointing to another template,
polycom.xml).
   res phoneprov also registers its own dialplan function: PP EACH USER.
This function was designed to be able to print out a particular string for each
user that res phoneprov knows about. An example use of this function is the
template for a Polycom contact directory:

  <?xml version="1.0" standalone="yes"?>
  <directory>
    <item_list>
      ${PP_EACH_USER(<item><fn>%{DISPLAY_NAME}</fn><ct>%{CALLERID}</ct><bw>1</bw></item>|${MAC})}
    </item_list>
  </directory>
   PP EACH USER takes two arguments. The first is the string to be
printed for each user. Any variables that are to be substituted need to be
in the format %{VARNAME} so that Asterisk doesn't try to substitute the
variable immediately before it is passed to PP EACH USER. The second,
optional, argument is a MAC address to exclude from the list iterated over
(so, in this case, a phone won't be listed in its own contact directory).


                                             197


14.5 Putting it all together

Make sure that manager.conf has:

  [general]
  enabled = yes
  webenabled = yes

   and that http.conf has:
  [general]
  enabled = yes
  bindaddr = 192.168.1.1 ; Your IP here ;-)
  bindport = 8088 ; Or port 80 if it is the only http server running on the machine

   With phoneprov.conf and users.conf in place, start Astersik. From
the CLI, type "http show status". An example output:
  HTTP Server Status:
  Prefix: /asterisk
  Server Enabled and Bound to 192.168.1.1:8088

  Enabled URI's:
  /asterisk/httpstatus => Asterisk HTTP General Status
  /asterisk/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
  /asterisk/manager => HTML Manager Event Interface
  /asterisk/rawman => Raw HTTP Manager Event Interface
  /asterisk/static/... => Asterisk HTTP Static Delivery
  /asterisk/mxml => XML Manager Event Interface
  Enabled Redirects:
    None.
  POST mappings:
  None.

   There should be a phoneprov URI listed. Next, from the CLI, type
"phoneprov show routes" and verify that the information there is correct.
An example output for Polycom phones woud look like:
  Static routes

  Relative URI Physical location
  sip.ver configs/sip.ver
  sip.ld configs/sip.ld
  bootrom.ver configs/bootrom.ver
  sip.cfg configs/sip.cfg
  bootrom.ld configs/bootrom.ld
  custom.cfg configs/custom.cfg
  Dynamic routes


                                           198


  Relative URI Template
  deadbeef4dad.cfg 000000000000.cfg
  deadbeef4dad-directory.xml 000000000000-directory.xml
  deadbeef4dad-phone.cfg 000000000000-phone.cfg
  config/deadbeef4dad polycom.xml
   With the above examples, the phones would be pointed to http://192.168.1.1:8080/asterisk/phoneprov
for pulling config files. Templates would all be placed in AST DATA -
DIR/phoneprov and static files would be placed in AST DATA DIR/phoneprov/configs.
Examples of valid URIs would be:

   * http://192.168.1.1:8080/asterisk/phoneprov/sip.cfg
   * http://192.168.1.1:8080/asterisk/phoneprov/deadbeef4dad.cfg

   * http://192.168.1.1:8080/asterisk/phoneprov/config/deadbeef4dad




















                                         199




Chapter 15

Packet Loss Concealment


15.1 What is PLC?
PLC stands for Packet Loss Concealment. PLC describes any method of
generating new audio data when packet loss is detected. In Asterisk, there
are two main flavors of PLC, generic and native. Generic PLC is a method
of generating audio data on signed linear audio streams. Signed linear audio,
often abbreviated "slin," is required since it is a raw format that has no
companding, compression, or other transformations applied. Native PLC
is used by specific codec implementations, such as iLBC and Speex, which
generates the new audio in the codec's native format. Native PLC happens
automatically when using a codec that supports native PLC. Generic PLC
requires specific configuration options to be used and will be the focus of this
document.

15.2 How does Asterisk detect packet loss?

Oddly, Asterisk does not detect packet loss when reading audio in. In order
to detect packet loss, one must have a jitter buffer in use on the channel
on which Asterisk is going to write missing audio using PLC. When a jitter
buffer is in use, audio that is to be written to the channel is fed into the
jitterbuffer. When the time comes to write audio to the channel, a bridge
will request that the jitter buffer gives a frame of audio to the bridge so that
the audio may be written. If audio is requested from the jitter buffer but the

                                          200


jitter buffer is unable to give enough audio to the bridge, then the jitter buffer
will return an interpolation frame. This frame contains no actual audio data
and indicates the number of samples of audio that should be inserted into
the frame.

15.3 A bit of background on translation
As stated in the introduction, generic PLC can only be used on slin audio.
The majority of audio communication is not done in slin, but rather using
lower bandwidth codecs. This means that for PLC to be used, there must
be a translation step involving slin on the write path of a channel. This
means that PLC cannot be used if the codecs on either side of the bridge
are the same or do not require a translation to slin in order to translate
between them. For instance, a ulaw <-> ulaw call will not use PLC since no
translation is required. In addition, a ulaw <-> alaw call will also not use
PLC since the translation path does not include any step involving slin. One
item of note is that slin must be present on the write path of a channel since
that is the path where PLC is applied. Consider that Asterisk is bridging
channels A and B. A uses ulaw for audio and B uses GSM. This translation
involves slin, so things are shaping up well for PLC. Consider, however if
Asterisk sets up the translation paths like so:

                            Fig. 1

A +------------+ B
<---ulaw<---slin<---GSM| |GSM--->
                                 | Asterisk |
ulaw--->slin--->GSM--->| |<---GSM
                                 +------------+

The arrows indicate the direction of audio flow. Each channel has a write
path (the top arrow) and a read path (the bottom arrow). In this setup, PLC
can be used when sending audio to A, but it cannot be used when sending
audio to B. The reason is simple, the write path to A's channel contains a slin
step, but the write path to B contains no slin step. Such a translation setup
is perfectly valid, and Asterisk can potentially set up such a path depending

                                           201


on circumstances. When we use PLC, however, we want slin audio to be
present on the write paths of both A and B. A visual representation of what
we want is the following:

                             Fig. 2
A +------------+ B
<---ulaw<---slin| |slin--->GSM--->
                        | Asterisk |
ulaw--->slin--->| |<---slin<---GSM
                        +------------+


In this scenario, the write paths for both A and B begin with slin, and so
PLC may be applied to either channel. This translation behavior has, in the
past been doable with the transcode_via_sln option in asterisk.conf.
Recent changes to the PLC code have also made the genericplc option in
codecs.conf imply the transcode_via_sln option. The result is that by
enabling genericplc in codecs.conf, the translation path set up in Fig. 2
should automatically be used.

15.4 Additional restrictions and caveats
One restriction that has not been spelled out so far but that has been hinted
at is the presence of a bridge. The term bridge in this sense means two
channels exchanging audio with one another. A bridge is required because
use of a jitter buffer is a prerequisite for using PLC, and a jitter buffer is
only used when bridging two channels. This means that one-legged calls,
(e.g. calls to voicemail, to an IVR, to an extension that just plays back
audio) will not use PLC. In addition, MeetMe and ConfBridge calls will not
use PLC. It should be obvious, but it bears mentioning, that PLC cannot be
used when using a technology's native bridging functionality. For instance,
if two SIP channels can exchange RTP directly, then Asterisk will never be
able to process the audio in the first place. Since translation of audio is a
requirement for using PLC, and translation will not allow for a native bridge
to be created, this is something that is not likely to be an issue, though. Since
a jitter buffer is a requirement in order to use PLC, it should be noted that

                                           202


simply enabling the jitter buffer via the jbenable option may not be enough.
For instance, if bridging two SIP channels together, the default behavior will
not be to enable jitter buffers on either channel. The rationale is that the
jitter will be handled at the endpoints to which Asterisk is sending the audio.
In order to ensure that a jitter buffer is used in all cases, one must enable
the jbforce option for channel types on which PLC is desired.

15.5 Summary
The following are all required for PLC to be used:

    * Enable genericplc in the plc section of codecs.conf
    * Enable (and potentially force) jitter buffers on channels
    * Two channels must be bridged together for PLC to be used (no Meetme
       or one-legged calls)
    * The audio must be translated between the two channels and must have
       slin as a step in the translation process.

15.6 Protip

One of the restrictions mentioned is that PLC will only be used when two
audio channels are bridged together. Through the use of Local channels, you
can create a bridge even if the call is, for all intents and purposes, one-legged.
By using a combination of the /n and /j suffixes for a Local channel, one can
ensure that the Local channel is not optimized out of the talk path and that
a jitter buffer is applied to the Local channel as well. Consider the following
simple dialplan:

[example]
exten => 1,1,Playback(tt-weasels)
exten => 2,1,Dial(Local/1@example/nj)

When dialing extension 1, PLC cannot be used because there will be only
a single channel involved. When dialing extension 2, however, Asterisk will
                                             203


create a bridge between the incoming channel and the Local channel, thus
allowing PLC to be used.




























                                   204




Chapter 16

Sounds Packages


16.1 Introduction

Asterisk utilizes a variety of sound prompts that are available in several file
formats and languages. Multiple languages and formats can be installed on
the same system, and Asterisk will utilize prompts from languages installed,
and will automatically pick the least CPU intensive format that is available
on the system (based on codecs in use, in additional to the codec and format
modules installed and available).
   In addition to the prompts available with Asterisk, you can create your
own sets of prompts and utilize them as well. This document will tell you
how the prompts available for Asterisk are created so that the prompts you
create can be as close and consistent in the quality and volume levels as those
shipped with Asterisk.

16.2 Getting The Sounds Tools
The sounds tools are available in the publicly accessible repotools repository.
You can check these tools out with Subversion via the following command:
  # svn co http://svn.asterisk.org/svn/repotools
   The sound tools are available in the subdirectory sound tools/ which
contains the following directories:
   * audiofilter

                                          205


   * makeg722
   * scripts

16.3 About The Sounds Tools

The following sections will describe the sound tools in more detail and explain
what they are used for in the sounds package creation process.

16.3.1 audiofilter
The audiofilter application is used to "tune" the sound files in such a way that
they sound good when being used while in a compressed format. The values
in the scripts for creating the sound files supplied in repotools is essentially
a high-pass filter that drops out audio below 100Hz (or so).
   (There is an ITU specification that states for 8KHz audio that is being
compressed frequencies below a certain threshold should be removed because
they make the resulting compressed audio sound worse than it should.)
   The audiofilter application is used by the 'converter' script located in the
scripts subdirectory of repotools/sound tools. The values being passed to
the audiofilter application is as follows:
  audiofilter -n 0.86916 -1.73829 0.86916 -d 1.00000 -1.74152 0.77536
   The two options -n and -d are 'numerator' and 'denominator'. Per the
author, Jean-Marc Valin, "These values are filter coefficients (-n means nu-
merator, -d is denominator) expressed in the z-transform domain. There
represent an elliptic filter that I designed with Octave such that 'the result
sounds good'."

16.3.2 makeg722
The makeg722 application is used by the 'converters' script to generate the
G.722 sound files that are shipped with Asterisk. It starts with the RAW
sound files and then converts them to G.722.



                                             206


16.3.3 scripts
The scripts folder is where all the magic happens. These are the scripts that
the Asterisk open source team use to build the packaged audio files for the
various formats that are distributed with Asterisk.

   * chkcore - used to check that the contents of core-sounds-<lang>.txt
     are in sync
   * chkextra - same as above, but checks the extra sound files
   * mkcore - script used to generate the core sounds packages

   * mkextra - script used to generate the extra sounds packages
   * mkmoh - script used to generate the music on hold packages
   * converters - script used to convert the master files to various formats


















                                        207




Chapter 17

Development



17.1 Backtrace
This document is intended to provide information on how to obtain the back-
traces required on the asterisk bug tracker, available at http://bugs.digium.com.
The information is required by developers to help fix problem with bugs of
any kind. Backtraces provide information about what was wrong when a
program crashed; in our case, Asterisk. There are two kind of backtraces
(aka 'bt') which are useful: bt and bt full.
   First of all, when you start Asterisk, you MUST start it with option -g.
This tells Asterisk to produce a core file if it crashes.
   If you start Asterisk with the safe asterisk script, it automatically starts
using the option -g.
   If you're not sure if Asterisk is running with the -g option, type the
following command in your shell:
  debian:/tmp# ps aux | grep asterisk
  root 17832 0.0 1.2 2348 788 pts/1 S Aug12 0:00 /bin/sh /usr/sbin/safe_asterisk
  root 26686 0.0 2.8 15544 1744 pts/1 S Aug13 0:02 asterisk -vvvg -c
  [...]

   The interesting information is located in the last column.
   Second, your copy of Asterisk must have been built without optimization
or the backtrace will be (nearly) unusable. This can be done by selecting
the 'DONT OPTIMIZE' option in the Compiler Flags submenu in the 'make
menuselect' tree before building Asterisk.

                                            208


   After Asterisk crashes, a core file will be "dumped" in your /tmp/ direc-
tory. To make sure it's really there, you can just type the following command
in your shell:
  debian:/tmp# ls -l /tmp/core.*
  -rw------- 1 root root 10592256 Aug 12 19:40 /tmp/core.26252
  -rw------- 1 root root 9924608 Aug 12 20:12 /tmp/core.26340
  -rw------- 1 root root 10862592 Aug 12 20:14 /tmp/core.26374
  -rw------- 1 root root 9105408 Aug 12 20:19 /tmp/core.26426
  -rw------- 1 root root 9441280 Aug 12 20:20 /tmp/core.26462
  -rw------- 1 root root 8331264 Aug 13 00:32 /tmp/core.26647
  debian:/tmp#

   In the event that there are multiple core files present (as in the above
example), it is important to look at the file timestamps in order to determine
which one you really intend to look at.
   Now that we've verified the core file has been written to disk, the final
part is to extract 'bt' from the core file. Core files are pretty big, don't be
scared, it's normal.
   NOTE: Don't attach core files on the bug tracker, we only need
the bt and bt full.
   For extraction, we use a really nice tool, called gdb. To verify that you
have gdb installed on your system:
  debian:/tmp# gdb -v
  GNU gdb 6.3-debian
  Copyright 2004 Free Software Foundation, Inc.
  GDB is free software, covered by the GNU General Public License, and you are
  welcome to change it and/or distribute copies of it under certain conditions.
  Type "show copying" to see the conditions.
  There is absolutely no warranty for GDB. Type "show warranty" for details.
  This GDB was configured as "i386-linux".
  debian:/tmp#

   Which is great, we can continue. If you don't have gdb installed, go install
gdb.
   Now load the core file in gdb, as follows:
  debian:/tmp# gdb asterisk /tmp/core.26252
  [...]
  (You would see a lot of output here.)
  [...]
  Reading symbols from /usr/lib/asterisk/modules/app_externalivr.so...done.
  Loaded symbols for /usr/lib/asterisk/modules/app_externalivr.so
  #0 0x29b45d7e in ?? ()
  (gdb)

   Now at the gdb prompt, type: bt You would see output similar to:

                                              209


(gdb) bt
#0 0x29b45d7e in ?? ()
#1 0x08180bf8 in ?? ()
#2 0xbcdffa58 in ?? ()
#3 0x08180bf8 in ?? ()
#4 0xbcdffa60 in ?? ()
#5 0x08180bf8 in ?? ()
#6 0x180bf894 in ?? ()
#7 0x0bf80008 in ?? ()
#8 0x180b0818 in ?? ()
#9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180
#10 0x000000a0 in ?? ()
#11 0x000000a0 in ?? ()
#12 0x00000000 in ?? ()
#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
#15 0xbcdffbe0 in ?? ()
#16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
#17 0x401ec92a in clone () from /lib/libc.so.6
(gdb)

 The bt's output is the information that we need on the bug tracker.

Now do a bt full as follows:
(gdb) bt full
#0 0x29b45d7e in ?? ()
No symbol table info available.
#1 0x08180bf8 in ?? ()
No symbol table info available.
#2 0xbcdffa58 in ?? ()
No symbol table info available.
#3 0x08180bf8 in ?? ()
No symbol table info available.
#4 0xbcdffa60 in ?? ()
No symbol table info available.
#5 0x08180bf8 in ?? ()
No symbol table info available.
#6 0x180bf894 in ?? ()
No symbol table info available.
#7 0x0bf80008 in ?? ()
No symbol table info available.
#8 0x180b0818 in ?? ()
No symbol table info available.
#9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180
No locals.
#10 0x000000a0 in ?? ()
No symbol table info available.
#11 0x000000a0 in ?? ()
No symbol table info available.
#12 0x00000000 in ?? ()
No symbol table info available.
#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
        f = (struct ast_frame *) 0x8180bf8
        trans = (struct ast_trans_pvt *) 0x0
#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
No locals.

                                         210


   #15 0xbcdffbe0 in ?? ()
   No symbol table info available.
   #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
   No symbol table info available.
   #17 0x401ec92a in clone () from /lib/libc.so.6
   No symbol table info available.
   (gdb)

    We also need gdb's output. That output gives more details compared to
the simple "bt". So we recommend that you use bt full instead of bt. But,
if you could include both, we appreciate that.
    The final "extraction" would be to know all traces by all threads. Even
if asterisk runs on the same thread for each call, it could have created some
new threads.
    To make sure we have the correct information, just do: (gdb) thread
apply all bt
   Thread 1 (process 26252):
   #0 0x29b45d7e in ?? ()
   #1 0x08180bf8 in ?? ()
   #2 0xbcdffa58 in ?? ()
   #3 0x08180bf8 in ?? ()
   #4 0xbcdffa60 in ?? ()
   #5 0x08180bf8 in ?? ()
   #6 0x180bf894 in ?? ()
   #7 0x0bf80008 in ?? ()
   #8 0x180b0818 in ?? ()
   #9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180
   #10 0x000000a0 in ?? ()
   #11 0x000000a0 in ?? ()
   #12 0x00000000 in ?? ()
   #13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 "DAHDI/pseudo-1324221520") at app_meetme.c:262
   #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
   #15 0xbcdffbe0 in ?? ()
   #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
   #17 0x401ec92a in clone () from /lib/libc.so.6
   (gdb)

    That output tells us crucial information about each thread.
    Now, just create an output.txt file and dump your "bt full" (and/or "bt")
ALONG WITH "thread apply all bt" into it.
    Note: Please ATTACH your output, DO NOT paste it as a note.
    And you're ready for upload on the bug tracker.
    If you have questions or comments regarding this documentation, feel free
to pass by the #asterisk-bugs channel on irc.freenode.net.



                                             211


