[asterisk-bugs] [Asterisk 0010868]: 1.4.11 Stable - Polycom phones hang up when media is re-invited while resuming from an on-hold state

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A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10868 
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Reported By:                mavince
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10868
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.11  
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:             10-02-2007 09:23 CDT
Last Modified:              10-05-2007 11:37 CDT
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Summary:                    1.4.11 Stable - Polycom phones hang up when media is
re-invited while resuming from an on-hold state
Description: 
PSTN calls made with Polycom phones (several different firmware loads) hang
up when resuming the call from a hold state. 

The call flow connects a Polycom phone to a PSTN phone through a SONUS
gateway with the media passing directly from to/from the Polycom phone. 

Call Scenario: The call can be initiated and answered in either direction,
to or from the PSTN. RTP is successfully provided. If the Polycom phone end
places the call on hold, MOH will be heard. If the Polycom phone then
attempts to resume the call, Asterisk issues two nearly simultaneous
INVITEs with an incremented CSeq, resulting in a 491 - Request Pending (the
appropriate response). Asterisk acknowledges the 491 and then hangs up the
call! I can reproduce the call behavior consistently.

Key points: media is passing directly to Polycom phone, call can be placed
on-hold successfully, other phones Snom, Aastra work correctly with the
same configuration. When media traverses the Asterisk, a Polycom based call
works correctly.

See Bug Tracker Issue 0009921
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Relationships       ID      Summary
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related to          0009921 1.4.4 sends Re-INVITE twice, resulting ...
related to          0009431 Modify connection: Response 491 not han...
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---------------------------------------------------------------------- 
 mavince - 10-05-07 11:37  
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I had forgot about the different Hold options. My initial settings were:
SIP voIpProt.SIP.useRFC2543hold="0" voIpProt.SIP.useSendonlyHold="1"

I tried different combinations with no change in results. Using the
RFC2543
I did see the media attribute change to "inactive" instead of "sendonly"

My sip.conf has

nat=no
canreinvite=yes

My call scenario calls to/from an IP Phone behind the Asterisk to/from 
a PSTN based phone.

I can make the same calls through a Cisco 2811 and get the same response
as I observed with the Sonus GSX gateway

Attached WireShark trace, Asterisk sends two identical INVITES (number 17
& 19 in trace) to the Session Border Controller...that invokes the behavior
associated with the hangup 

Issue History 
Date Modified   Username       Field                    Change               
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10-05-07 11:37  mavince        Note Added: 0071539                          
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