| Summary: | ASTERISK-09249: G729 codec lock up when build with Jitter buffer | ||
| Reporter: | Saket Setu (saketsetu) | Labels: | |
| Date Opened: | 2007-04-12 07:01:58 | Date Closed: | 2011-06-07 14:03:20 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/CodecInterface |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Compiled asterisk with Jitter buffer patch enabled. http://svn.digium.com/svn/asterisk/team/oej/jitterbuffer-1.2/ Logged in an Agent in a Queue. Initiating an outgoung call using Action Originate command to another asterisk m/c and bridging the call to agent. The SIP Gateway supports Codec is g729. Number of codecs available is 10. The first call is bridged and codec in use is 1. Used Action: Redirect command to move the call to Meetme Room. Audio becomes one way and error on asterisk is Out of G.729 Encoder Licenses!. show g729 gives an output 10/0 (encoder/decoder) in use. Rebuild the same code without JB and every thing is fine. Followed few posts http://bugs.digium.com/view.php?id=3854 http://lists.digium.com/pipermail/asterisk-dev/2005-April/011015.html Is it anything to do with G729 codec not supporting jitter? | ||
| Comments: | By: Joshua C. Colp (jcolp) 2007-04-12 10:43:56 We don't support branches as such like this. Please get in contact with oej and see if it's still up to date and meant for usage (I doubt it). | ||