[asterisk-bugs] [Asterisk 0010567]: DTMF INFO event appears to be causing Maximum retries exceeded on transmission hangup.

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10567 
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Reported By:                jacksch
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   10567
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:            1.4.9  
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:             08-26-2007 23:09 CDT
Last Modified:              01-31-2008 13:28 CST
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Summary:                    DTMF INFO event appears to be causing Maximum
retries exceeded on transmission hangup.
Description: 
I've spent a *lot* of time struggling with outgoing calls being
disconnected 20 seconds after they were answered.  Since my phones are on a
LAN (private addressing) and my Asterisk box is dual homed (one interface
with a public IP), there is definately no NAT problem. My Asterisk talks
SIP to all my local devices on the LAN, and sends outbound calls for
termination via IAX2.

If I dial a number and hit pound (to cause the call to be dialled
immediately) on a phone connected to my Sipura 2000, Asterisk connects the
call, and I have two way audio, but then it begins transmitting INVITES to
the ATA.  The ATA acknowledging them, but, it still results in :

[Aug 27 00:01:02] WARNING[15661]: chan_sip.c:1920 retrans_pkt: Maximum
retries exceeded on transmission 9d4091f7-a13e2f8d at 192.168.220.52 for seqno
102 (Critical Response)
[Aug 27 00:01:02] WARNING[15661]: chan_sip.c:1944 retrans_pkt: Hanging up
call 9d4091f7-a13e2f8d at 192.168.220.52 - no reply to our critical packet.

If I place the same call and just wait for the ATA to time out (as opposed
to hitting # after dialling the number), the problem does not occur.

Please contact me if you would like the full debug file -- eric( a t
)jacksch( d o t )com
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---------------------------------------------------------------------- 
 oej - 01-31-08 13:28  
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Please try with latest 1.4 as we've changed a lot of functionality in
regards to in-dialog messages. If it still doesn't work, please upload a
full SIP debug including debug output at the level of 5. (Make sure debug
channel is enabled in logger.conf).

THanks. 

Issue History 
Date Modified   Username       Field                    Change               
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01-31-08 13:28  oej            Note Added: 0081516                          
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