bender*CLI> sip set debug peer 880 SIP Debugging Enabled for IP: 192.168.8.2:5060 -- Accepting AUTHENTICATED call from 200.174.44.76: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (alaw|gsm), > priority = mine -- Executing [889@ramais:1] Dial("IAX2/central-7", "SIP/880|30|Ttr") in new stack Audio is at 192.168.8.1 port 14142 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 192.168.8.2:5060: INVITE sip:880@192.168.8.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895;rport From: "Flavio do Carmo Junior" ;tag=as5d5f06ee To: Contact: Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Jul 2007 18:00:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 260 v=0 o=root 3328 3328 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 14142 RTP/AVP 0 8 101 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 --- -- Called 880 bender*CLI> <--- SIP read from 192.168.8.2:5060 ---> SIP/2.0 100 Trying To: From: "Flavio do Carmo Junior" ;tag=as5d5f06ee Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- bender*CLI> <--- SIP read from 192.168.8.2:5060 ---> SIP/2.0 180 Ringing To: ;tag=3af6fd51637e6f1ei0 From: "Flavio do Carmo Junior" ;tag=as5d5f06ee Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/880-082080a8 is ringing bender*CLI> <--- SIP read from 192.168.8.2:5060 ---> SIP/2.0 200 OK To: ;tag=3af6fd51637e6f1ei0 From: "Flavio do Carmo Junior" ;tag=as5d5f06ee Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK737a8895 Contact: 880 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 229 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 65980 65980 IN IP4 192.168.8.2 s=- c=IN IP4 192.168.8.2 t=0 0 m=audio 16450 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (12 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.8.2:16450 Found description format PCMU for ID 0 Found description format NSE for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.8.2:16450 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.8.2, port 5060 Transmitting (no NAT) to 192.168.8.2:5060: ACK sip:880@192.168.8.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK396d2ed0;rport From: "Flavio do Carmo Junior" ;tag=as5d5f06ee To: ;tag=3af6fd51637e6f1ei0 Contact: Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/880-082080a8 answered IAX2/central-7 [Jul 10 15:00:24] WARNING[3338]: chan_iax2.c:7760 socket_process: Received mini frame before first full voice frame <--- SIP read from 192.168.8.2:5060 ---> BYE sip:402@192.168.8.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.2:5060;branch=z9hG4bK-5a618fcb From: ;tag=3af6fd51637e6f1ei0 To: "Flavio do Carmo Junior" ;tag=as5d5f06ee Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 101 BYE Max-Forwards: 70 User-Agent: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.8.2 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.8.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.8.2:5060;branch=z9hG4bK-5a618fcb;received=192.168.8.2 From: ;tag=3af6fd51637e6f1ei0 To: "Flavio do Carmo Junior" ;tag=as5d5f06ee Call-ID: 2cb94a57528739a343de46b759ef9e7f@192.168.8.1 CSeq: 101 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (ramais, 889, 1) exited non-zero on 'IAX2/central-7' -- Hungup 'IAX2/central-7' Really destroying SIP dialog '2cb94a57528739a343de46b759ef9e7f@192.168.8.1' Method: BYE bender*CLI>