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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15337 
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Reported By:                deitch
Assigned To:                dvossel
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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:              
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Date Submitted:             2009-06-16 17:49 CDT
Last Modified:              2009-06-26 17:00 CDT
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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.
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 (0107076) dvossel (administrator) - 2009-06-26 17:00
 https://issues.asterisk.org/view.php?id=15337#c107076 
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I'd give 1.4.25.1 a shot out of curiosity.  Changing to a different version
really isn't that big of a deal.  Just clean out the modules, probably in
usr/lib/asterisk/modules/, between installations.  Your etc/asterisk
configs should stay the same unless you do a 'make samples'

Also, you didn't address my other question, does voice sound right, or is
it just prerecordings that are messed up? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-26 17:00 dvossel        Note Added: 0107076                          
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