Summary:ASTERISK-05818: SIP reinvite and realtime peers
Reporter:alterys (alterys)Labels:
Date Opened:2005-12-12 03:58:58.000-0600Date Closed:2011-06-07 14:03:20
Versions:Frequency of
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
OS: CentOS-4
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.