|Summary:||ASTERISK-05818: SIP reinvite and realtime peers|
|Date Opened:||2005-12-12 03:58:58.000-0600||Date Closed:||2011-06-07 14:03:20|
|Environment:||Attachments:||( 0) debug|
|Description:||Reinvite doesn't work with realtime peers.|
* If 2 realtime peers communicate, asterisk stays in the media path, sound is ok
* If one realtime peer communicate with a static one, media path also transits by asterisk
* If 2 static peers communicate, everything is fine, media path directly going from one peer to the other.
It seems that realtime peers doesn't care abour the canreinvite parameter.
Version: asterisk 1.2-beta1
|Comments:||By: Kevin P. Fleming (kpfleming) 2005-12-13 09:40:37.000-0600|
As requested in the bug posting guidelines, we cannot do _anything_ to help you without a complete console trace showing the problem occurring.
By: alterys (alterys) 2005-12-13 10:18:14.000-0600
Here is a debug file containing call establishment between two SJPhone peers on a LAN, same IP range, same codec (GSM). 5 seconds of call duration.
By: Kevin P. Fleming (kpfleming) 2005-12-13 10:25:57.000-0600
Your dial string contains the 'Tt' options; with transfer enabled, reinvite cannot occur. This is a configuration problem.