[Asterisk-bugs] [Asterisk 0009921]: 1.4.4 sends Re-INVITE twice, resulting in code 491 "Request pending" and call termination by Asterisk

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email_notification_title_for_action_bugnote_submitted 
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http://bugs.digium.com/view.php?id=9921 
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Reported By:                fabianhoppe
Assigned To:                
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Project:                    Asterisk
Issue ID:                   9921
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.4 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        No 
Request Review:              
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Date Submitted:             06-08-2007 04:42 CDT
Last Modified:              07-03-2007 07:55 CDT
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Summary:                    1.4.4 sends Re-INVITE twice, resulting in code 491
"Request pending" and call termination by Asterisk
Description: 
Dear All,

I'm using Asterisk v1.4.4 with the lastet patch level (as of 27.04.07).
Several Polycom phones in a remote location are connected to the Asterisk
server via the Internet with IPCoP and siproxd 0.5.13 serving as outbound
proxy. All outbound traffic is routed via SIP to a local SIP provider,
running a TELES iSwitch as softswitch. My sip.conf is listed below under
"additional information". In this setup I run into the following,
reproducable problem:

When I place an outbound call, the call setup works fine and the reinvites
take place, RTP streams a setup between the Polycom and the SIP provider
directly. 

Placing the call on HOLD then results in a re-invite to the SIP-provider
who answers with "100 trying". 

Immediately Asterisk resends an identical re-invite (with an incremented
CSeq), resulting in a "491 Request Pending" from the SIP provider (this is
the correct answer, as the first INVITE is not yet fully processed). 

This gets acknowledged by Asterisk and then Asterisk simply terminates the
call!! 

You can find the SIP DEBUG output from the Asterisk server as attachment.
I can provide a wireshark capture on request. Please let me know, if more
data / logs are needed.
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---------------------------------------------------------------------- 
 fabianhoppe - 07-03-07 07:55  
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Gentlemen,

I couldn't reproduce the bug described above - neither with the original
version nor with the svn trunk. Please have a look at the attached pcap
around packet no 246 to see that the bug really exists.

We decided to use reinvites for internal calls and to disable reinvites on
SIP trunks.

Please close this bug report, bug report 9305 probably covers this issue
too.

Best Regards, Fabian Hoppe 

Issue History 
Date Modified   Username       Field                    Change               
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07-03-07 07:55  fabianhoppe    Note Added: 0066448                          
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