[asterisk-bugs] [Asterisk 0017372]: Progress in band error (don't send RTP packets)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17372 
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Reported By:                tech_admin
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17372
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.7 
JIRA:                       SWP-1526 
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:             2010-05-21 09:31 CDT
Last Modified:              2010-06-01 21:22 CDT
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Summary:                    Progress in band error (don't send RTP packets)
Description: 
Dear all,

on a updated redhat rhel5 (kernel: 2.6.18-194.3.1.el5xen), I installed the
last asterisk release (1.6.2.7) from the source code.

The problem is: SIP caller doesn't hear ringing tone.
We use progressinband=yes with ch (Switzerland) tones and voice is G729
encoded using TC400B Digium card.
After answer, all is ok: both side RTP packets go trough asterisk without
translation.
The Digium card work fine with others service like IVR, Queues, Voicemail
... all using G729.


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Relationships       ID      Summary
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related to          0017185 [patch] [regression] Using Local channe...
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 (0122747) DennisD (reporter) - 2010-06-01 21:22
 https://issues.asterisk.org/view.php?id=17372#c122747 
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I just tested with 1.6.2.9-rc1.

I used the above example, except I used extension 520 instead of 6111.

I've attached issue-17372-no_ringing_debug.log that was when
progressinband=yes was set in sip.conf and
issue-17372-with_ringing_debug.log after I commented it out and did a sip
reload.

I've also attached issue-17372.pcap showing the RTP packets are there, but
I couldn't hear anything.

Oddly, when I tested with a Future-Nine DID, Asterisk did NOT send out any
RTP packets to Future-Nine (I tested with Progress() then
Dial(SIP/xxxx,30)), BUT when I tested with a voip.ms DID, I heard the early
media.

Let me know what else you want and I'll attach it as soon as I can get
back to this. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-06-01 21:22 DennisD        Note Added: 0122747                          
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