[asterisk-bugs] [Asterisk 0011740]: DTMF problem on 1.4.17

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11740 
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Reported By:                gserra
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11740
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.17 
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:             01-11-2008 05:55 CST
Last Modified:              01-17-2008 10:13 CST
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Summary:                    DTMF problem on 1.4.17
Description: 
Hi,

when a call is being passed over from AS5400 via PRI to SIP into asterisk,
dtmf is working in 1.4.11 but not on 1.4.17
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---------------------------------------------------------------------- 
 aragon - 01-17-08 10:13  
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Here is Snom's response..
1. Does your system (asterisk) receive these SIP INFO messages (when you
check in its trace (because they are certainly being sent out).
2. If so, why is the system then not acting on them?

r it could be the SIP INFO messages are being sent too quickly for the
system (but that should not be the case since its SIP messages). If you
could confirm the above two questions we can investigate further. We
appreciate your help.
Thanks.

Sent to tls:192.168.0.86:5061 at 21/12/2007 17:59:15:656 (477 bytes):
 
 INFO sip:250 at 192.168.0.86:5061;transport=tls SIP/2.0
 Via: SIP/2.0/TLS 192.168.10.93:5060;branch=z9hG4bK-sf9wxiqyhuym;rport
 From: "250 PBX" <sip:250 at localhost>;tag=hdp3x0uy7w
 To: <sip:251 at localhost;user=phone>;tag=3541c0b07b
 Call-ID: acf06b47b240-zwf7xn8780xc
 CSeq: 6 INFO
 Max-Forwards: 70
 Contact:
<sip:250 at 192.168.10.93:5060;transport=tls;line=21tq03ix>;flow-id=1
 User-Agent: snom370/
 Content-Type: application/dtmf-relay
 Content-Length: 22
 
 Signal=7
 Duration=160
 
 

--------------------------------------------------------------------------------

 
 Sent to tls:192.168.0.86:5061 at 21/12/2007 17:59:15:671 (477 bytes):
 
 INFO sip:250 at 192.168.0.86:5061;transport=tls SIP/2.0
 Via: SIP/2.0/TLS 192.168.10.93:5060;branch=z9hG4bK-uixxv9owokm9;rport
 From: "250 PBX" <sip:250 at localhost>;tag=hdp3x0uy7w
 To: <sip:251 at localhost;user=phone>;tag=3541c0b07b
 Call-ID: acf06b47b240-zwf7xn8780xc
 CSeq: 7 INFO
 Max-Forwards: 70
 Contact:
<sip:250 at 192.168.10.93:5060;transport=tls;line=21tq03ix>;flow-id=1
 User-Agent: snom370/
 Content-Type: application/dtmf-relay
 Content-Length: 22
 
 Signal=8
 Duration=160
 
 

--------------------------------------------------------------------------------

 
 Sent to tls:192.168.0.86:5061 at 21/12/2007 17:59:15:671 (477 bytes):
 
 INFO sip:250 at 192.168.0.86:5061;transport=tls SIP/2.0
 Via: SIP/2.0/TLS 192.168.10.93:5060;branch=z9hG4bK-yl2d4hdhmows;rport
 From: "250 PBX" <sip:250 at localhost>;tag=hdp3x0uy7w
 To: <sip:251 at localhost;user=phone>;tag=3541c0b07b
 Call-ID: acf06b47b240-zwf7xn8780xc
 CSeq: 8 INFO
 Max-Forwards: 70
 Contact:
<sip:250 at 192.168.10.93:5060;transport=tls;line=21tq03ix>;flow-id=1
 User-Agent: snom370/
 Content-Type: application/dtmf-relay
 Content-Length: 22
 
 Signal=6
 Duration=160


I have uploaded asteriskSIPinfoDTMFfail.log
Maybe the answer is in the Asterisk trace
If the trace is not sufficient please let me know and I will provide
another trace with more options set. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-17-08 10:13  aragon         Note Added: 0080811                          
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