hostname*CLI> <--- SIP read from Asterisk:5060 ---> INVITE sip:some_ext@Asterisk:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0 Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564 Content-Length: 355 Contact: Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126 Content-Type: application/sdp CSeq: 2 INVITE From: "unknown";tag=2127514061631 Max-Forwards: 69 To: v=0 o=- 3388311921 3388311921 IN IP4 192.168.0.126 s=SJphone c=IN IP4 Asterisk t=0 0 a=direction:active m=audio 37924 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 a=nortpproxy:yes <-------------> --- (12 headers 16 lines) --- Sending to Asterisk : 5060 (no NAT) Using INVITE request as basis request - 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126 Found no matching peer or user for 'Asterisk:5060' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port Asterisk:37924 Found description format GSM for ID 3 Found description format iLBC for ID 97 Found description format iLBC for ID 98 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port Asterisk:37924 Looking for some_ext in default (domain Asterisk) list_route: hop: hostname*CLI> <--- Transmitting (no NAT) to Asterisk:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564 From: "unknown";tag=2127514061631 To: Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [some_ext@default:1] Answer("SIP/5060-00663300", "") in new stack Audio is at Asterisk port 11176 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to Asterisk:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564 Record-Route: From: "unknown";tag=2127514061631 To: ;tag=as082880aa Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 285 v=0 o=root 15109 15109 IN IP4 Asterisk s=session c=IN IP4 Asterisk t=0 0 m=audio 11176 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [some_ext@default:2] Playback("SIP/5060-00663300", "tt-weasels") in new stack -- Playing 'tt-weasels' (language 'en') Retransmitting #1 (no NAT) to Asterisk:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564 Record-Route: From: "unknown";tag=2127514061631 To: ;tag=as082880aa Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 285 v=0 o=root 15109 15109 IN IP4 Asterisk s=session c=IN IP4 Asterisk t=0 0 m=audio 11176 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to Asterisk:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564 Record-Route: From: "unknown";tag=2127514061631 To: ;tag=as082880aa Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 285 v=0 o=root 15109 15109 IN IP4 Asterisk s=session c=IN IP4 Asterisk t=0 0 m=audio 11176 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv