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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Asterisk PBX functions

Implementing common PBX functions with Asterisk


Some PBX functions in Asterisk are implemented as applications or are supported by a combination of applications that you use in the dial plan. Next to that the different channels implement different subsets of CLASS (or Vertical Service) codes.

general support (for all channels)

  • Music on Hold: Implemented in Asterisk. See MusicOnHold and musiconhold.conf.
  • Call Parking: Supported in the standard installation
  • Call Pickup: Supported in the standard installation (*8 - defined in res_features.c +55, modify pickupexten in features.conf)
  • Call Recording: Using the 'Monitor' application
  • Conferencing: Using the 'MeetMe' application
  • IVR: Supported in extensions.conf through the Asterisk applications; employ AGI or EAGI if even more control is needed
  • DISA: Direct Inwards System Access, Allows someone from outside the telephone switch (PBX) to obtain an "internal" system dialtone.
  • Voicemail: Voicemail System, For unavailable or busy or no answer status, call can be automatic routed to Voicemail system. Voicemail system has many more attractive features.

for SIP Phones

  • Call Hold: Normally implemented by your phone
  • Unattended Transfer (or "blind transfer"): Implemented in Asterisk (#), optionally also in the phone
  • Consultation Hold: Normally implemented by your phone, for
  • Unconditional Call Forwarding
  • Attended Transfer (or "consultative transfer")
  • No Answer Call Forwarding: Implemented by yourself in the dial plan. See the tips & tricks page for ideas
  • Busy Call Forwarding:Implemented by yourself in the dial plan. See the tips & tricks page for ideas
  • Single-Line Extension:
  • 3-way Calling: Normally implemented by the phone
  • Incoming Call Screening: Implemented by yourself in the dial plan
  • Find-Me:
  • Call Pickup: Supported in the standard installation
  • Outgoing Call Screening: Implemented by yourself in the dial plan
  • Automatic Redial: You should be able to implement this in the dial plan with some AGI support
  • Manual Redial
  • Do-not-disturb (DND)
  • Message waiting (MWI): Implemented in Asterisk, but must be support on the phone
  • Call waiting indication: Implemented in Asterisk, but must be support on the phone






for Analogue Phones connect to Zaptel channel

See Asterisk vertical service activation codes for ZAP channels
  • Call Hold: Normally implemented by your phone
  • Unattended Transfer (or "blind transfer")
  • Consultation Hold: Normally implemented by your phone, for
  • Unconditional Call Forwarding
  • Attended Transfer (or "consultative transfer"): See Asterisk tips zap transfer
  • No Answer Call Forwarding: Implemented by yourself in the dial plan. See the tips & tricks page for ideas
  • Busy Call Forwarding:Implemented by yourself in the dial plan. See the tips & tricks page for ideas
  • Single-Line Extension:
  • 3-way Calling: Normally implemented by the phone
  • Incoming Call Screening: Implemented by yourself in the dial plan
  • Find-Me:
  • Call Pickup: Supported in the standard installation
  • Outgoing Call Screening: Implemented by yourself in the dial plan
  • Automatic Redial: You should be able to implement this in the dial plan with some AGI support
  • Manual Redial

  • Do-not-disturb (DND)
  • Message waiting (MWI): Implemented in Asterisk, but must be support on the phone


.....Improved by Nahid Hossain

for MGCP Phones

See Asterisk MGCP channels
  • Manual Redial: Normally implemented by your phone
  • Unattended transfer (or "blind transfer"): Implemented in Asterisk (#)
  • Attended transfer: Implemented in Asterisk (FLASH)
  • Call Forwarding: Implemented in Asterisk (*72 and *73); optionally implemented in the phone
  • Call Pickup: Implemented in Asterisk (*8)
  • Call Waiting Indication: Implemented in Asterisk; disable with *70
  • Call Number Delivery Blocking: Implemented in Asterisk (*67)
  • Do-not-disturb (DND): Normally implemented by your phone; also implemented in Asterisk (*78 and *79)
  • Message waiting (MWI): Implemented in Asterisk, but must be support on the phone

on the CAPI channel

See Asterisk CAPI channels
  • Call Deflection (CD) (redirect without answering): Implemented by chan_capi
  • CLIP & CLIR (display caller ID & hide my caller ID): Implemented by chan_capi
  • CID & DNID: Implemented by chan_capi
  • HOLD & RETRIEVE: Hold a call using ISDN (not the PBX): Implemented by chan_capi
  • Early B3 Connects (always,success,never): Implemented by chan_capi
  • DID (for Point to Point mode): Implemented by chan_capi

  • ECT (explicit call transfer): Preserve the orginial CID - Implemented by chan_capi


This page is now beginning to be more complete in any way./OEJ

See also

Created by oej, Last modification by Paul Gillman on Tue 24 of Apr, 2007 [18:23 UTC]

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