<--- SIP read from UDP://79.131.24.197:5060 ---> INVITE sip:8899@212.90.154.15;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.27.175:5060;branch=z9hG4bK-d8754z-d4f192d4c2911584-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;transport=UDP From: "12345";transport=UDP;tag=0e5f0f56 Call-ID: MmY1ZjNiODM5MjI2MzVhNWUzOGFiZjgxYjI3NTkzNjU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO Content-Type: application/sdp User-Agent: Zoiper for Windows rev.1105 Content-Length: 210 v=0 o=Zoiper_user 0 0 IN IP4 192.168.27.175 s=Zoiper_session c=IN IP4 192.168.27.175 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 10 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.27.175 : 5060 (no NAT) Using INVITE request as basis request - MmY1ZjNiODM5MjI2MzVhNWUzOGFiZjgxYjI3NTkzNjU. Found peer '12345' for '12345' from 79.131.24.197:5060 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.27.175:8000 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=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.27.175:8000 Looking for 8899 in athineou (domain 212.90.154.15) list_route: hop: WinAster3*CLI> <--- Transmitting (NAT) to 79.131.24.197:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.27.175:5060;branch=z9hG4bK-d8754z-d4f192d4c2911584-1---d8754z-;received=79.131.24.197;rport=5060 From: "12345";transport=UDP;tag=0e5f0f56 To: ;transport=UDP Call-ID: MmY1ZjNiODM5MjI2MzVhNWUzOGFiZjgxYjI3NTkzNjU. CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer ontact: Content-Length: 0 <------------> -- Executing [8899@athineou:1] Set("SIP/12345-0847a738", "_dialerid=") in new stack -- Executing [8899@athineou:2] Set("SIP/12345-0847a738", "_voice=") in new stack -- Executing [8899@athineou:3] Hangup("SIP/12345-0847a738", "") in new stack == Spawn extension (athineou, 8899, 3) exited non-zero on 'SIP/12345-0847a738' Scheduling destruction of SIP dialog 'MmY1ZjNiODM5MjI2MzVhNWUzOGFiZjgxYjI3NTkzNjU.' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 79.131.24.197:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.27.175:5060;branch=z9hG4bK-d8754z-d4f192d4c2911584-1---d8754z-;received=79.131.24.197;rport=5060 From: "12345";transport=UDP;tag=0e5f0f56 To: ;transport=UDP;tag=as0e5ecaf5 Call-ID: MmY1ZjNiODM5MjI2MzVhNWUzOGFiZjgxYjI3NTkzNjU. CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> WinAster3*CLI> <--- SIP read from UDP://79.131.24.197:5060 ---> ACK sip:8899@212.90.154.15;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.27.175:5060;branch=z9hG4bK-d8754z-d4f192d4c2911584-1---d8754z-;rport To: ;transport=UDP;tag=as0e5ecaf5 From: "12345";transport=UDP;tag=0e5f0f56 Call-ID: MmY1ZjNiODM5MjI2MzVhNWUzOGFiZjgxYjI3NTkzNjU. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) ---