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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 tips answer-before-playback

John Todd, of loligo.com advises:

  • Before running any application that has sound playback (Playback, Background, VoiceMailMain2, etc.) it would be wise to execute an Answer first, then a Wait(2) to allow for VoIP channels to fully establish and settle.

Wasim adds:

Instead of a wait(2), we generally background an innocous music or a prompt for say 2 seconds, thus, people aren't waiting in dead-air, get some audio-feedback, and if the prompt is truncated, it doesn't matter since it was just a feeder in the first place. Wait will also not take any DTMF input during that time, so repeat users are stuck for that time.

Ofcourse if you notice a significant % of your VoIP sessions take time to settle, then best do a Wait(2), no point in send garble-jumble down the line. JT, as usual has very valuable suggestions.

Note and update

New versions of Asterisk have added "Answer" capabilities to several functions like Playback(), which means that those functions will answer themselves if necessary.

C F, shmaltz at gmail dot com:
I was having problems that zap (FXO) channels were answered when I assumed that they will not be answered because the context didn't have any answer cmd in it. However I found out (the hard way), that you DONT have to do answer before using playback, and even without answer the ZAP line gets answered when using playback.
You can find it all here:
http://lists.digium.com/pipermail/asterisk-users/2005-January/082966.html
I know that under Note and update someone already posted it, but since I wasn't the only one in the list that didn't notice this note and update I decided to add this.


Back to Asterisk tips and tricks
Created by oej, Last modification by shmaltz on Sun 16 of Jan, 2005 [18:27 UTC]

Comments Filter

Re: SPA-941 DND and wait...

by Ian on Wednesday 04 of January, 2006 [01:40:12 UTC]
Sounds like you just need to do

exten => you,1,Dial(SIP/you|20)
exten => you,2,Voicemail(uyou)
exten => you,3,Hangup()
exten => you,102,Wait(2)
exten => you,103,Voicemail(byou)
exten => you,104,Hangup()

SPA-941 DND and wait...

by Scott Brown on Wednesday 04 of January, 2006 [01:01:49 UTC]
How would this work for my SPA-941... When I press the DND button, calls go directly to VM.. BUT, the first 2 seconds of my message get sucked off... Anyone know how to change this?

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