[asterisk-bugs] [Asterisk 0016625]: RFC2833 DTMF is not passed correctly when going SIP->IAX2->SIP

Asterisk Bug Tracker noreply at bugs.digium.com
Sun Jan 17 22:09:34 CST 2010


The following issue NEEDS LICENSE. 
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https://issues.asterisk.org/view.php?id=16625 
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Reported By:                sharvanek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16625
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     needs license
Asterisk Version:           1.4.28 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-01-17 12:02 CST
Last Modified:              2010-01-17 22:09 CST
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Summary:                    RFC2833 DTMF is not passed correctly when going
SIP->IAX2->SIP
Description: 
We were the victims of the Sonus bug, we've patched appropriately however
we're still having an interesting issue.

SIP->Asterisk->SIP->Sonus works fine now for DTMF via RFC2833 using the
attached patch.

However if you toss IAX2 in the middle (even with both asterisk boxes
patched) DTMF is unreliable, (jitterbuffer is off).

SIP Phone->Asterisk->IAX2->Asterisk->SIP->Sonus

Attached are packet captures from the last asterisk box in the path to
Sonus for both SIP (working) and IAX2 (not working).

If we substitute the IAX2 trunk for SIP (still using the two asterisk
boxes) DTMF works flawlessly, something is getting lost when going over
IAX2 and we'd like to figure out what that is and get it corrected.

Thanks

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Issue History 
Date Modified    Username       Field                    Change               
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2010-01-17 22:09 snuffy         Status                   new => needs license
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