<--- SIP read from UDP:192.168.1.172:19643 ---> INVITE sip:123456@192.168.1.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.172:19643;branch=z9hG4bK0600131e9e From: "ata-s1" ;tag=3f401166 To: Call-ID: 60babae6034483a7071f4a9e175bceb9@192.168.1.172 Contact: CSeq: 1 INVITE Max-Forwards: 70 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK Allow-Events: talk,hold,conference Supported: replaces,100rel Content-Type: application/sdp User-Agent: CM5K-TA1S (807100.02) Content-Length: 302 v=0 o=CMI-SIPUA 20287 0 IN IP4 192.168.1.172 s=SIP CALL c=IN IP4 192.168.1.172 t=0 0 m=audio 20026 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 a=rtcp:20027 a=sendrecv <-------------> --- (14 headers 15 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.1.172 : 19643 (no NAT) Using INVITE request as basis request - 60babae6034483a7071f4a9e175bceb9@192.168.1.172 No matching peer for '1122334' from '192.168.1.172:19643' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.172:20026 Looking for 123456 in defctxt (domain 192.168.1.110) list_route: hop: <--- Transmitting (no NAT) to 192.168.1.172:19643 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.172:19643;branch=z9hG4bK0600131e9e;received=192.168.1.172 From: "ata-s1" ;tag=3f401166 To: Call-ID: 60babae6034483a7071f4a9e175bceb9@192.168.1.172 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [123456@defctxt:1] Hangup("SIP/192.168.1.110-00000000", "") in new stack == Spawn extension (defctxt, 123456, 1) exited non-zero on 'SIP/192.168.1.110-00000000' Scheduling destruction of SIP dialog '60babae6034483a7071f4a9e175bceb9@192.168.1.172' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 192.168.1.172:19643 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.172:19643;branch=z9hG4bK0600131e9e;received=192.168.1.172 From: "ata-s1" ;tag=3f401166 To: ;tag=as5035cd0e Call-ID: 60babae6034483a7071f4a9e175bceb9@192.168.1.172 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:192.168.1.172:19643 ---> ACK sip:123456@192.168.1.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.172:19643;branch=z9hG4bK0600131e9e From: "ata-s1" ;tag=3f401166 To: ;tag=as5035cd0e Call-ID: 60babae6034483a7071f4a9e175bceb9@192.168.1.172 Contact: CSeq: 1 ACK Max-Forwards: 70 User-Agent: CM5K-TA1S (807100.02) Content-Length: 0 <-------------> --- (10 headers 0 lines) ---