search for: audiowriteformat

...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"

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:

...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 t.38. Is there a way to get t...

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).