login | register
Wed 01 of Aug, 2007 [09:46 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • 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
Server Stats
  • Execution time: 0.38s
  • Memory usage: 2.19MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 2.77

NVLineDetect

Answer Detection, Dialtone Detection, and Dead Channel Detection for ZAP and IAX, SIP, others

Including busy detect, congestion detection, and ring detection. Contact Newman Telecom for the code.

Synopsis

 Detects answer/dead/other signals on ZAP and other channels

Description

 NVLineDetect([waitdur[|options[|deaddur[|sildur[|mindur[|maxdur]]]]]])

This application listens for certain tones (on ZAP and most channel types) for max waitdur seconds of time. In particular, it detects presence of conditions: ANSWER, PICKUP, RING, BUSY, CONGESTION, DIALTONE, and DEAD. The respective extension name (in lowercase letters) will be called (i.e. 'answer', 'pickup', etc.). In addition, the variable TONE_DETECTED is set. If all undetected, control will continue at the next priority.

Parameters

     waitdur:  Maximum number of seconds to wait (default=30)
     options:
       'n':  Attempt on-hook if unanswered (default=no)
       'd':  Ignore answer detection of ring+talk (default=no)
       'a':  Ignore pickup detection of ring+sil (default=no)
       'b':  Ignore busy detection (default=no)
       'c':  Ignore congestion detection (default=no)
       'r':  Return after ring/ringing detection (default=no)
       'd':  Return after dialtone detection (default=no)
       's':  Ignore dead channel detection (default=no)
     deaddur:  How many ms of no activity to wait for dead (default=30)
     sildur:  Silence ms after min/maxdur before answer/pickup (default=1000)
     mindur:  Minimum non-silence talk ms needed (default=100)
     maxdur:  Maximum non-silence talk ms allowed (default=0/forever)

Return codes

Returns -1 on hangup, and 0 on successful completion with no exit conditions.

Notes

This code is NOT included with Asterisk at this point, however it is free. To get it, e-mail Newman Telecom at jnewman@newmantelecom.com.. We'll respond quickly.

This code works best on ZAP channel or channels using ULAW/ALAW, however it will work with other codecs.

Requirements

  • Asterisk development or stable

Sample Usage (extensions.conf)

[context-incoming]
; Answer and do some detection work
exten => s,1,Answer
exten => s,2,NVLineDetect
exten => s,3,Hangup

; The channel is dead
exten => dead,1,Hangup

; Play welcome, send to primary
exten => answer,1,NVBackgroundDetect(welcome)
exten => answer,2,Dial(SIP/5500)
exten => answer,2,Hangup

; If this is a fax, dial fax line
exten => fax,1,Dial(SIP/5501)
exten => fax,2,Hangup

; If user is talking, send him to Debra
exten => talk,1,Dial(SIP/5502)
exten => talk,2,Hangup

Installation

Easiest way to get up and running:

(1) Drop the code in your /usr/src/asterisk/apps directory

(2) Edit the Makefile in the apps directory. Add the following line:
    APPS+=app_nv_linedetect.so

(3) Go to /usr/src/asterisk and run "make", then run "make install"

(4) Start or restart Asterisk

(5) Type "show application nvlinedetect" from the CLI and you should see it

NOTE: Dialtone detection requires additional changes. This will be posted shortly.


Future Improvements

None at this time.

Compatability

The copy I obtained included instructions to replace the existing dsp.c dsp.h and frame.h files with the ones supplied. This causes asterisk 1.2.9 errors when compiling.
Why isn't a patch supplied when existing files need to be modified?
Copying a modified version of source code from an older version of asterisk into a current version could introduce a number of bugs.

See also


Created by justin_newman, Last modification by Steve Switzer on Mon 29 of Jan, 2007 [04:12 UTC]

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver