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Wed 01 of Aug, 2007 [09:18 UTC]

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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 PSTN interface debugging

When interfacing between Asterisk and the PSTN many problems can arise, you can debug your PSTN interface.


Echo


Signal Level

  • ztmonitor <channel number> -v
    • gives you a visual representation of the sound strengths and makes it easy to see if the receive or transmit signals are too high or out of balance

Randomly Dropped Calls

  • Read comments in zapata.conf for the callprogress setting.
  • Try setting busydetect=no or greater than 6

How to speak Telco

Linemen and installers have to debug wierd line situations all the time. Accordingly they've built testing facilities on to specific lines so they can debug echo and so on. Take a look at http://www.nettwerked.net/K-1ine_37.txt. Pay careful attention to Quiet Termination and Milliwatt test lines - you will need to get friendly with a telco field person to find out the access numbers at your serving Central Office.

_Nahid

Asterisk

Created by jht2, Last modification by nahid on Tue 02 of Aug, 2005 [17:27 UTC]

Comments Filter

what command to see the line ringing

by sjobeck on Wednesday 05 of January, 2005 [09:00:03 UTC]
From the CLI, what command to see the ZAP channels ring?

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