[asterisk-bugs] [Asterisk 0012615]: DTMF transmission is randomly ignored during SIP-SIP calls

[noreply at bugs.digium.com]

The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=12615 
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Reported By:                adeel
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12615
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             05-09-2008 00:30 CDT
Last Modified:              05-13-2008 15:02 CDT
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Summary:                    DTMF transmission is randomly ignored during SIP-SIP
calls
Description: 
It seems that Asterisk is randomly ignoring DTMF generated from the locally
connected Polycom 601's/650's/330's. This problem occurs randomly
irrespective of which ITSP I use or what number I call. Furthermore, one
digit maybe ignored while the next is passed through (in the same call).
I've also experienced DTMF troubles even when checking voicemail, so I
don't think it's a network/connectivity issue. The DTMF logs indicate that
the digit was received but ignored (doesn't mention why, maybe we should
put in more descriptive error messages?). 

I'm running Asterisk 1.4.18, and have applied a DTMF patch i found on this
bugzilla relating to DTMF and bridging calls, which seems to have helped,
but not totally solve the problem. The device & user peer definition is set
to use RFC2833, while sip.conf defaults to using AUTO and relaxdtmf IS
currently set in sip.conf.

it seems that one of the if tests on line 2156 of channels.c is failing,
but not sure which one. I'm wondering if Polycom handles DTMF in a
non-standard way, or something. I tried testing with another vendor phone
(all i had was a grandstream) and still had mixed results (some calls
handled DTMF just fine, a few calls wouldn't recognize the DTMF).

I've had this problem ever since I switched to 1.4 (might have had it in
1.2, but it was never as prominent/problematic as it is now), and I'm close
to my wits end on the problem.

Unfortunately I don't have the sip debug messages, but do have the CLI
verbose output and the DTMF logs, so I'm including them below.
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---------------------------------------------------------------------- 
 file - 05-13-08 15:02  
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Please use an unpatched version of Asterisk with rfc2833compensate set to
yes in the general section and report back. If this does not solve the
issue please attach (as a file) complete console log output including rtp
debug. 

Issue History 
Date Modified   Username       Field                    Change               
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05-13-08 15:02  file           Note Added: 0086810                          
05-13-08 15:02  file           Status                   new => feedback     
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