[asterisk-bugs] [Asterisk 0013265]: RFC2833 mangled from Sonus when RTP stream passes through asterisk rather than being reinvited

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 8 10:26:58 CST 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13265 
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Reported By:                ptimmins
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13265
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21.2 
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:             2008-08-09 17:00 CDT
Last Modified:              2008-12-08 10:26 CST
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Summary:                    RFC2833 mangled from Sonus when RTP stream passes
through asterisk rather than being reinvited
Description: 
We were/are using 1.4.11. We connect to Global Crossing over SIP. They use
Sonus softswitches. We use ulaw and rfc2833. We also use Cisco AS5400
gateways and Adtran Total Access 900 IADs. Under 1.4.11, RFC2833 DTMF
passed through asterisk was properly replicated. When we upgraded
(directly) to 1.4.21.2, asterisk began manipulating the DTMF such that when
it went from Global, through asterisk, to the Cisco or Adtran gateways, the
touchtones were mangled to the point they simply make a chirping silence
noise. If canreinvite=yes is set, they work fine. However, that Global
doesn't allow you to reinvite to arbitrary destinations, so this won't work
as a long term solution. Attached is a copy of a working SIP+RTP stream
going from Global to Asterisk, and from Asterisk to the as5400, as well as
a copy of a failing one, as well as the dtmf debug.
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---------------------------------------------------------------------- 
 (0095955) file (administrator) - 2008-12-08 10:26
 http://bugs.digium.com/view.php?id=13265#c95955 
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I need to see your configuration from sip.conf for this plus complete
console output. The RFC2833 digits are getting sent with a different
timestamp/sequence number generation then the audio which is causing things
to get screwed up... this should not happen if everything is going through
the core as it is should for this so it may be some sort of weird
configuration that is causing the two channels to Packet2Packet bridge when
they should not. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-08 10:26 file           Note Added: 0095955                          
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