search for: audionativeformat

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 thr...

...looks like: [default] exten => _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL}) exten => _X.,n,Dial(SIP/user at domain.net,30,M(getCalledInfo)) exten => h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})}) [macro-getCalledInfo] exten => s,1,Set(SHARED(X-DST-CODEC,${X-SRC_CHANNEL})=${CHANNEL(audionativeformat)}) The above works great, however there is a problem when call is transferred via SIP attended transfer and channel is renamed to Channel<ZOMBIE>. -- Executing [h at default:1] Set("SIP/somechannel<ZOMBIE>", "CDR(DST_CODEC)=") in new stack Is there any workaround...

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:

Sorry if this is an obvious question and perhaps my Google foo isn't right on this one: I have calls coming into an Asterisk server that may be using 2 different codecs. I am recording audio in both cases but the challenge is knowing which codec was negotiated at call setup. I need to pass the proper format to the record command as the codecs cannot be transcoded and are only supported 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

...one press the pound key. <SIP/1320-00000188>AGI Tx >> agi_arg_2: en <SIP/1320-00000188>AGI Tx >> <SIP/1320-00000188>AGI Rx << CHANNEL STATUS <SIP/1320-00000188>AGI Tx >> 200 result=6 <SIP/1320-00000188>AGI Rx << GET FULL VARIABLE ${CHANNEL(audionativeformat)} <SIP/1320-00000188>AGI Tx >> 200 result=1 ((gsm)) -- <SIP/1320-00000188>AGI Script googletts.agi completed, returning 0 -- Executing [1310 at Client-dial-Menu:2] Hangup("SIP/1320-00000188", "") in new stack == Spawn extension (Client-dial-Menu, 1310...

...atistics 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 t.38. Is there a way to get the codec for both legs of a ca...