Im facing strange issue while establishing inbound calls from SIP trunks. Provider A is sending (G729, Alaw, uLaw) offer and asterisk dial the peer with its preferred codec order(G729, aLaw, uLaw). The peers phone send the codec list as (uLaw, spe..
Im facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says *Dial requi..
Try MixMonitor. Land the call to a local channel and answer it. This code will record the silence as well.exten => _X.,1,MixMonitor()exten => _X.,n,Dial(Local/100@context1)[context1]exten => _X.,1,Answer()exten => _X.,n,Dial(SI..
Im observing wrong From/Contact header values. When I try to set CallerID(num) it has no effect in the From and Contact Headers, and these values are the same as the dialed number. SIP Peers are defined using asterisk realtime. If I define the SIP Pe..
is possible that two sip extensions: user-1 and user-2 are connected and Iwant to transfer the call from user-1 to a third user user-3. I know it is possible through feature keys mapping in features.conf, but Iwant to do this through AMI or Aster..
Im getting an issue while executing AMI Originate. Im getting extension does not exists on Originates Response, and on the other hand Asterisk CLI say fwrite() returned error: Broken pipePlease suggest me what is wrong.Muhammad Faheem### my origin..