Hello,<br>I'm wrong or it should be a=sendonly in 183 Session Progress ?<br><br>Calling an application Playback with flag 'noanswer' generates 183 with a=sendrecv.<br><br>Here is a sip debug of 183:<br><--- Transmitting (no NAT) to <a href="http://192.168.245.25:5060">192.168.245.25:5060</a> ---><br>
SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP 192.168.245.25:5060;branch=z9hG4bK7da3300e;received=192.168.245.25;rport=5060<br>From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as2fe7eeee<br>To: <sip:user01@sip.testserver.lan>;tag=as7993bb65<br>
Call-ID: <a href="mailto:0616052b6bb4b4a45897ad1e706cc98f@192.168.245.25">0616052b6bb4b4a45897ad1e706cc98f@192.168.245.25</a><br>CSeq: 102 INVITE<br>Server: Asterisk PBX 1.8.11-cert7<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Contact: <<a href="http://sip:user01@192.168.245.10:5060">sip:user01@192.168.245.10:5060</a>><br>Content-Type: application/sdp<br>Content-Length: 290<br><br>v=0<br>o=root 1550475731 1550475731 IN IP4 192.168.245.10<br>
s=Asterisk PBX 1.8.11-cert7<br>c=IN IP4 192.168.245.10<br>t=0 0<br>m=audio 13218 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>
a=ptime:20<br>a=sendrecv<br><br><br><br>