search for: _sip_codec_outbound

...andles per minute and it never goes down. The dialplan is actually a four liner look at the audiowritecodec select an outbound endpoint based on that The idea is to bridge calls based on the codec to avoid any transcoding, so I have two outbound codecs and I dial like this: exten => _X.,1,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audiowriteformat):0:4}) exten => _X.,n,Goto(${SIP_CODEC_OUTBOUND}) exten => _X.,n(ulaw),Dial(PJSIP/alawoutbound/sip:${EXTEN}@X.X.X.X) exten => _X.,n(g729),Dial(PJSIP/g729outbound/sip:${EXTEN}@X.X.X.X) As you can see, "Houston, we have a problem"

...appear to be able to set an inheritable variable for the subsequent PJSIP leg of the call, to exclusively only offer the codec we negotiated for the first leg of the call. If for example we have chan_iax2 incoming that we wish to send out via pjsip. With chan_sip, this works: exten => s,n,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audioreadformat)}) With pjsip, this gives an error: exten => s,n,Set(_PJSIP_MEDIA_OFFER(audio)=!all,${CHANNEL(audioreadformat)}) Error: ERROR[26925][C-00020b9c] pbx_functions.c: Function _PJSIP_MEDIA_OFFER not registered I'd image things haven't changed since 2018 where...