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

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=9921 
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Reported By:                fabianhoppe
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   9921
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
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:              
Resolution:                 unable to reproduce
Fixed in Version:           
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Date Submitted:             06-08-2007 04:42 CDT
Last Modified:              07-05-2007 08:27 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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 file - 07-05-07 08:27  
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Closed per request, please reopen if this creeps up again. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-05-07 08:27  file           Status                   feedback => resolved
07-05-07 08:27  file           Resolution               open => unable to
reproduce
07-05-07 08:27  file           Assigned To               => file            
07-05-07 08:27  file           Note Added: 0066521                          
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