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  • Samuel, Thu 03 of Jul, 2008 [13:41 UTC]: ok thank you
  • Mats Karlsson, Thu 03 of Jul, 2008 [13:37 UTC]: Nice Samuel, will look forward to rad it.
  • bwl_fernstudent, Thu 03 of Jul, 2008 [09:08 UTC]: Your blog shows some usefull code
  • Samuel, Thu 03 of Jul, 2008 [08:04 UTC]: I'll translate it, for sure
  • Mats Karlsson, Wed 02 of Jul, 2008 [20:46 UTC]: LOL, in french! Translate it to English and I will read it.
  • Samuel, Wed 02 of Jul, 2008 [08:07 UTC]: Hello, i wrote a blog about Asterisk, speaking about installation,programming and more http://sambranche.blogspot.com/
  • Nick Barnes, Tue 01 of Jul, 2008 [17:46 UTC]: Steve - Asterisk doesn't 'fit into linux' - it's an application which runs on top of Linux.
  • Steve, Mon 30 of Jun, 2008 [18:07 UTC]: anyone know where I can find a block diagram of how asterisk fits into linux. my f'ing bosses want me to draw something up.. ugh.
  • akbar, Fri 27 of Jun, 2008 [10:37 UTC]: marley_boyz@yahoo.com how to configure call forward, call back, call pick up using TDM and asterisk 1.2.13... please help me.. thx...
  • Matthew Williams, Tue 24 of Jun, 2008 [22:37 UTC]: We are looking for Tier II VoIP Support Technicians in St Louis. Send resumes to mwilliams AT voxitas DOT com.
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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]

Comments Filter

Asteriskwin32 source code required

by SAQIB IRSHAD on Saturday 19 of January, 2008 [20:49:45 UTC]
Hi,
I need Asteriskwin32 source code. But i not been able to open ftp.asterisk.org site. plz help me.

Regards,
Saqib

Asteriskwin32 needed

by SAQIB IRSHAD on Saturday 19 of January, 2008 [20:47:25 UTC]
Hi,
I need Asteriskwin32 source code. But i not been able to open ftp.asterisk.org site. plz help me.

Regards,
Saqib

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