| Summary: | ASTERISK-24778: OPUS codec not working with chan_pjsip | ||
| Reporter: | PowerPBX (PowerPBX) | Labels: | |
| Date Opened: | 2015-02-10 13:26:45.000-0600 | Date Closed: | 2015-02-10 14:29:50.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_pjsip |
| Versions: | 12.8.1 13.2.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | CentOS 6 64bit PJSIP v2.3 compiled from source using the asterisk recommendations Asterisk compiled from source with opus codec checked in make menuselect opus-devel installed from EPEL repo. opus codec added to asterisk source from https://github.com/seanbright/asterisk-opus | Attachments: | |
| Description: | With 2 chan_pjsip extensions set for OPUS codec I cannot make echo test or extension to extension calls. The extension will ring but the call is terminated as soon as I answer. Enabling pjsip debug logging was of no help. When I switch to speex codec I can then make calls. When I change the extensions to use chan_sip I can then make OPUS codec calls.
Unfortunately my setup goes through NAT so I was unable to verify if this occurs when using passhthrough for this codec. | ||
| Comments: | By: Matt Jordan (mjordan) 2015-02-10 14:29:50.337-0600 The {{codec_opus}} module is not part of Asterisk. If there are issues with it, you will need to contact the maintainers of that module. By: PowerPBX (PowerPBX) 2015-02-10 14:56:54.777-0600 If I verify that passthrough is not working will you re-open this issue? By: Alexander Traud (traud) 2015-11-06 07:49:00.083-0600 No need to re-open because the cause is the same as in ASTERISK-24779. | ||