Hello, we use the freepbx distro and run into a problem when trying to use speex.
In the case below i used eyebeam sip client with only the speex codec enabled and dialed the internal echo test application.
if i use g.711 it works just fine.
– Executing [*43@from-internal:2] Wait(“SIP/146-00001ff4”, “1”) in new stack
[2012-02-07 10:11:53] WARNING[21599]: channel.c:5103 set_format: Unable to find a codec translation path from 0x200 (speex) to 0x40 (slin)
[2012-02-07 10:11:53] ERROR[21599]: channel.c:7992 ast_channel_start_silence_generator: Could not set write format to SLINEAR
– Executing [*43@from-internal:3] Playback(“SIP/146-00001ff4”, “demo-echotest”) in new stack
[2012-02-07 10:11:54] WARNING[21599]: channel.c:5103 set_format: Unable to find a codec translation path from 0x200 (speex) to 0xe (gsm|ulaw|alaw)
[2012-02-07 10:11:54] WARNING[21599]: file.c:959 ast_streamfile: Unable to open demo-echotest (format 0x200 (speex)): No such file or directory
[2012-02-07 10:11:54] WARNING[21599]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/146-00001ff4 for demo-echotest
– Executing [*43@from-internal:4] Echo(“SIP/146-00001ff4”, “”) in new stack
First, I updated your post to format it correctly. It’s unreasonable to post unformatted output and expect anyone to study it. On this and any other forum if you take the time to provide and enough information to help and format it in an organized fashion you will get far more help.
Answer, the numbers are the transcoding delays. If no number is presented the CODEC is not installed. Asterisk will pass through any CODEC if it does not need to participate in the media stream. This is a very dicey option.
This blog discusses how to build CODEC’s against Asterisk source.