Re: [Linphone-users] Linphone opus and asterisk 13

[Anatoli]

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 <address@hidden>
Sent: Monday, July 08, 2019 09:26
To: Linphone-users <address@hidden>
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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