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Summary:ASTERISK-17503: Asterisk ignores reInvite
Reporter:aleksander2011 (aleksander2011)Labels:
Date Opened:2011-03-03 08:56:06.000-0600Date Closed:2011-06-07 14:05:12
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:1.6.2.17 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk.pcap
( 1) putty.log
Description:I do call from host1 (82.204.187.9) to Asterisk (192.168.0.28 - without NAT), and Asterisk sends this call to host2 (82.204.187.10). When customer is answering the call, then Asterisk connects RTP directly from host1 to host2. It work perfectly.
But If hostB wants send a reInvite to Asterisk (for example put on hold) and changes  media port, then the reinvite does't reach to host1 (lost in Asterisk). Asterisk received reInvite don't sent they in other leg. Why?
Option "canreinvite" set "yes" for all peer.
Comments:By: Leif Madsen (lmadsen) 2011-03-03 14:01:49.000-0600

Per the bug guidelines when filing SIP issues, please provide:

* SIP trace from the Asterisk console
* Configuration information and topology in order to make it possible to reproduce
* All devices being utilized in the topology
* Console output during the call setup

By: aleksander2011 (aleksander2011) 2011-03-04 01:58:06.000-0600

Hello lmadsen.

I have the following topology.
82.204.187.9 (Softswitch Essentra CX 4 class)
82.204.187.7 (Media Gateway)
82.204.187.10 (Softswitch Essentra BAX 5 class)
192.168.0.28 - Asterisk

82.204.187.9 -SIP->
                    192.168.0.28 -SIP+RTP->82.204.187.10 -SIP+RTP-> End User
82.204.187.7 -RTP->

I call to number +74956986015. End user +74956986015 is behind a 82.204.187.10. If he puts on hold incoming call or forwards call to another user then 82.204.187.10 changes RTP port and sends reinvite.
End user is behind a 82.204.187.10

I attached SIP trace from Asterisk console.
Asterisk received reinvites from 82.204.187.10 in line 1715 and 1888 (put on hold and forward), but Asterisk didn't send they to 82.204.187.9. And because of this after forward one way audio.



By: Leif Madsen (lmadsen) 2011-04-14 10:23:42

Honestly this feels like a support issue to me. Please use the asterisk-users mailing list to move this issue forward. If you can track down what the problem is then please reopen this issue and describe what Asterisk is doing wrong.