| Summary: | ASTERISK-05818: SIP reinvite and realtime peers | ||
| Reporter: | alterys (alterys) | Labels: | |
| Date Opened: | 2005-12-12 03:58:58.000-0600 | Date Closed: | 2011-06-07 14:03:20 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| 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. | ||