I think this is what your looking for: Found RTP audio format 119 Found audio description format speex for ID 119 Capabilities: us - (speex|speex16|speex32|g722|ulaw|alaw|gsm), peer - audio=(speex32)/video=(nothing)/text=(nothing), combined - (speex32) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.176:7078 [Jul 5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a codec translation path: (speex32) -> (speex) ^M^[[Kdevgeis*CLI> ^M^[[0K[Jul 5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a codec translation path: (speex) -> (speex32)
My linphone side only has speex@32K enabled. My extension definition has: disallow=all allow=speex allow=speex16 allow=speex32 allow=g722 allow=ulaw allow=alaw allow=gsm It looks like its the codec translation ? So then I enabled speex and speex32 on Linphone.... Got past that - I presume it will use speex32 for audio... But then I am trying to place that call in a conference (confbridge) and I get this error: Unable to find a codec translation path: (slin) -> (speex) so I think then it hangs up. What do I do about that ? - thanks Jerry On Fri, Jul 5, 2019 at 8:22 AM Jerry Geis <[email protected]> wrote: > Hi all - I am using asterisk 13.27.0 with Linphone. > I turned off all codes on linphone except the one I want to try. For > example: > opus and speex (so only one enabled at a time). > Then did this same on asterisk for the linphone extension. > disallow=all > allow=speex > > (for example). > > Then I place my call and the call fails. if I enable something like gsm, > ulaw, alaw the call works fine. Why does the call fail with opus and speex ? > Thanks, > > Jerry >
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