Hi Jerry,
The translation paths you show are only for 48KHz opus, but your
endpoints may use other rates. In any case you should enable logs and
see what's going on: pjsip set logger on.
Also, if both endpoints can use same rate opus, you may connect them
without transcoding.
Regards,
Anatoli
*From:* Jerry Geis <jerry.g...@gmail.com>
*Sent:* Monday, July 08, 2019 09:26
*To:* Linphone-users <linphone-users@nongnu.org>
*Subject:* [Linphone-users] Linphone opus and asterisk 13
I am trying to use linphone on windows with Asterisk 13. I set my
extension to use only opus, I set linphone to only use opus.
I followed the opus install for asterisk and put the codec in
/usr/lib/asterisk/modules.
Looks like the opus codec is installed:
core show translation paths opus
--- Translation paths SRC Codec "opus" sample rate 48000 ---
opus:48000 To g723:8000 : No Translation Path
opus:48000 To ulaw:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(ulaw@8000)
opus:48000 To alaw:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(alaw@8000)
opus:48000 To gsm:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(gsm@8000)
opus:48000 To g726:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(g726@8000)
opus:48000 To g726aal2:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(g726aal2@8000)
opus:48000 To adpcm:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(adpcm@8000)
opus:48000 To slin:8000 :
(opus@48000)->(slin@48000)->(slin@8000)
opus:48000 To slin:12000 :
(opus@48000)->(slin@48000)->(slin@12000)
opus:48000 To slin:16000 :
(opus@48000)->(slin@48000)->(slin@16000)
opus:48000 To slin:24000 :
(opus@48000)->(slin@48000)->(slin@24000)
opus:48000 To slin:32000 :
(opus@48000)->(slin@48000)->(slin@32000)
opus:48000 To slin:44100 :
(opus@48000)->(slin@48000)->(slin@44100)
opus:48000 To slin:48000 : (opus@48000)->(slin@48000)
opus:48000 To slin:96000 :
(opus@48000)->(slin@48000)->(slin@96000)
opus:48000 To slin:192000 :
(opus@48000)->(slin@48000)->(slin@192000)
opus:48000 To lpc10:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(lpc10@8000)
opus:48000 To g729:8000 : No Translation Path
opus:48000 To speex:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(speex@8000)
opus:48000 To speex:16000 :
(opus@48000)->(slin@48000)->(slin@16000)->(speex@16000)
opus:48000 To speex:32000 :
(opus@48000)->(slin@48000)->(slin@32000)->(speex@32000)
opus:48000 To ilbc:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(ilbc@8000)
opus:48000 To g722:16000 :
(opus@48000)->(slin@48000)->(slin@16000)->(g722@16000)
opus:48000 To siren7:16000 : No Translation Path
opus:48000 To siren14:32000 : No Translation Path
opus:48000 To testlaw:8000 :
(opus@48000)->(slin@48000)->(slin@8000)->(testlaw@8000)
opus:48000 To g719:48000 : No Translation Path
opus:48000 To none:8000 : No Translation Path
opus:48000 To silk:8000 : No Translation Path
opus:48000 To silk:12000 : No Translation Path
opus:48000 To silk:16000 : No Translation Path
opus:48000 To silk:24000 : No Translation Path
But when I call I call answers but I have no audio. What am I not doing ?
Thanks,
Jerry
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