search for: audioreadformat

...nheritable 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 this appears to have been...

Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that with ast-1.4.11. Thanks for ideas,...

...oked at this article about CDR in mysl. http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql I installed asterisk-mysql pacakge from debian repo. The cdr in mysql is working, but i can not get cdr adaptive colums are not, i use this in my extension.conf exten => s,1,set(CDR(teste)=${CHANNEL(audioreadformat)}) And is not working, i thought the only diference it i would need the colum teste in my cdr table right ? I tryied a lot of combinations of exten => Does anyone have any ideia how is the right way ? Or if i need to install anythign else to make the adaptive colums works ? thanks!!...

...rmation about CURRENT channel. But it won't allow me to obtain information about OTHER channels and that is what I want to do. I want a search for all channels and an output spit out as g729 or TRUE or FALSE if there is a g729 channel. exten => s,1,Answer() exten => s,n,Set(foo=${CHANNEL(audioreadformat)}) exten => s,n,NoOp(${foo}) Above ^^^^ NoOp spits out g729 if I call in with a g729 codec. But I want that to be about other channels and not the one I am calling into. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/piper...

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...s (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)} Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(audionativeformat)}/${SIPCHANINFO(t38passthrough)} QOS=${RTPAUDIOQOS}) The problems I have so far: *1. CODEC *Codec is reported only for A-Leg. When transcoding asterisk logs the above line as: slin for read / slin for write / the codec of A-Leg / 0 for...

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).