Summary:ASTERISK-06421: I have the following dial path, it places the call ok but does not pass audio
Reporter:Don Briggs (dbriggs54)Labels:
Date Opened:2006-02-25 10:10:05.000-0600Date Closed:2011-06-07 14:10:45
Versions:Frequency of
Environment:Attachments:( 0) asterisk1.txt
Description:Originated call to from outside PSTN t0 /sip/provider(boadvoice) > asterisk box 1 > forwared out IAX2 to asterisk box2  > then onto destiantion though PSDN.  Call completed fine but it passes no audio.  Both boxes are running asterisk 1.2.1 release.

Any ideas
Comments:By: Matt O'Gorman (mogorman) 2006-03-01 15:43:55.000-0600

we are gonna need a lot more info than this, if you could find me on irc mogorman, or over jabber mogorman@digium.com I'd love to help.  Can you perhaps issolate where in this call path the problem originates?

By: Tilghman Lesher (tilghman) 2006-03-11 09:47:26.000-0600

Any updates?

By: Martin Vit (festr) 2006-03-27 11:49:18.000-0600

it seems that this may be related to my issue 0006808.

dbriggs54: do you have jitterbuffer=yes and if so try to set jitterbuffer=no.

By: Jan Serve (jserve) 2006-04-03 16:28:03

I think I have the same problem. But my problem is that when I originate to a IAX2  Client I have this problem.

My enviroment sofar:
* Clients with several IAX2 Softphones (DIAX, IDEFISK and a own Application based on LIBIAX)
* Asterisk 1.2.6 on the life system, Asterisk SVN-trunk-r17151 on the testsystem.
* A Webinterface what connects via the Manager API and originates a call

Now why I add this note:
When I originate to exten *601(DIAL Attempt with Holding-Music) everything is working fine, when I originate to *602(without Holding-Music) I not hear anything.

Now something curios too, when I originate to *602 and press on IDEFISK the active line button a second time after few seconds I hear the opposite site as normal (and the opposite site hears me as normal).
Also when I debug my own develop application I see that there is coming in a second pickup request, when this is answered I hear also the opposite site(and can speak with it too).

I have attached a txt file(Ref.: asterisk1.txt) with some configuration information and the debug output of Asterisk.

With greetings Jan Serve

P.S.: I have try to change the jitterbuffer in the general section of iax.conf, but no change sofar.

By: Joshua C. Colp (jcolp) 2006-04-05 11:57:49

Set notransfer=yes in the peer and friend entries and see if the problem occurs again. Also - what trunk revision are you using? We had problems with multithreaded IAX2 which have been mostly solved on latest trunk. As well for problems like this, please don't use the issue tracker. It's meant for actual bug reports... yours is more like a request for help.

By: Jan Serve (jserve) 2006-04-05 14:46:20

notransfer=yes haven't work sofar, but I will try my luck on asterisk-users to not  make useless notes here.

By: Joshua C. Colp (jcolp) 2006-04-05 14:48:07

Problem is being moved to users list until it can be verified that this is an actual bug in Asterisk and not a misconfiguration, or issue with idefisk.