[asterisk-bugs] [Asterisk 0015337]: Drawn out and static audio on inbound iax2 calls

[Asterisk Bug Tracker]

The following issue has been set as RELATED TO issue 0015220. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15337 
====================================================================== 
Reported By:                deitch
Assigned To:                dvossel
====================================================================== 
Project:                    Asterisk
Issue ID:                   15337
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-06-16 17:49 CDT
Last Modified:              2009-07-01 08:55 CDT
====================================================================== 
Summary:                    Drawn out and static audio on inbound iax2 calls
Description: 
Calls that are inbound on a DID, to which Asterisk has registered using
iax2, have terribly drawn out audio and a lot of static for prerecorded
files (e.g. IVR and Voicemail files in /var/lib/asterisk/sounds). They
sound like:
".... hhhheeeellllllloooooo..... Yyyyyyoooouuuu hhhhaaaavvvveeee " etc.

Specific details:
1) This occurs only on IAX2, not on SIP. I have tried with the exact same
providers, on the same hosts
2) jitterbuffer is off. If I turn jitterbuffer on, no audio at all is
audible.
3) This occurs across two completely separate providers, one in Israel,
one in the US. However, inbound from the ITSPs directly to an IAX2
softphone (e.g. JackenIAX or Zoiper) work perfectly fine.
4) This occurs only when communicating to Asterisk. If the dialer enters
an extension, and is thus connected over Asterisk to a SIP softphone, there
is no problem.
5) This occurs independent of codec. I have the original outbound files
converted to both ulaw and gsm, and it occurs in both cases. The provider
prefers ulaw.
6) CPU usage by Asterisk remains extremely low during the bad output.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0015296 TDM call over IAX trunk gives choppy au...
related to          0015220 IAX2 choppy audio with MozIAX, only 20secs
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-01 08:55 lmadsen        Relationship added       related to 0015220  
======================================================================