pluto*CLI> sip debug peer gateway SIP Debugging Enabled for IP: 192.168.47.119:5060 pluto*CLI> pluto*CLI> Sending to 192.168.47.119 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.47.119:17718 Found description format PCMA Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 9999999 in remote list_route: hop: Transmitting (no NAT) to 192.168.47.119:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.47.119:5060 From: ;tag=DB685690-1170 To: Call-ID: BC98F3B6-D33311D9-85ED9E35-AB6E44EE@192.168.47.119 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/192.168.47.119-081a4d20", "SIP/2222222222@gateway&SIP/123|30|wW") in new stack We're at 192.168.47.101 port 13754 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.47.119:5060: INVITE sip:2222222222@192.168.47.119 SIP/2.0 Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK21adf880 From: "1111111111" ;tag=as1a8fe718 To: Contact: Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 06:58:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 21708 21708 IN IP4 192.168.47.101 s=session c=IN IP4 192.168.47.101 t=0 0 m=audio 13754 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 2222222222@gateway -- Called 123 pluto*CLI> <-- SIP read from 192.168.47.119:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK21adf880 From: "1111111111" ;tag=as1a8fe718 To: ;tag=DB6858D0-75E Date: Fri, 03 Jun 2005 06:58:35 GMT Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/123-6e9d is ringing Transmitting (no NAT) to 192.168.47.119:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.47.119:5060 From: ;tag=DB685690-1170 To: ;tag=as4ec92cda Call-ID: BC98F3B6-D33311D9-85ED9E35-AB6E44EE@192.168.47.119 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> <-- SIP read from 192.168.47.119:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK21adf880 From: "1111111111" ;tag=as1a8fe718 To: ;tag=DB6858D0-75E Date: Fri, 03 Jun 2005 06:58:35 GMT Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 185 v=0 o=CiscoSystemsSIP-GW-UserAgent 9753 7865 IN IP4 192.168.47.119 s=SIP Call c=IN IP4 192.168.47.119 t=0 0 m=audio 17916 RTP/AVP 8 c=IN IP4 192.168.47.119 a=rtpmap:8 PCMA/8000 --- (12 headers 8 lines)--- Found RTP audio format 8 Peer audio RTP is at port 192.168.47.119:17916 Found description format PCMA Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) -- SIP/gateway-6602 is making progress passing it to SIP/192.168.47.119-081a4d20 We're at 192.168.47.101 port 10568 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) Transmitting (no NAT) to 192.168.47.119:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.47.119:5060 From: ;tag=DB685690-1170 To: ;tag=as4ec92cda Call-ID: BC98F3B6-D33311D9-85ED9E35-AB6E44EE@192.168.47.119 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 21708 21708 IN IP4 192.168.47.101 s=session c=IN IP4 192.168.47.101 t=0 0 m=audio 10568 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> pluto*CLI> <-- SIP read from 192.168.47.119:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK21adf880 From: "1111111111" ;tag=as1a8fe718 To: ;tag=DB6858D0-75E Date: Fri, 03 Jun 2005 06:58:35 GMT Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 185 v=0 o=CiscoSystemsSIP-GW-UserAgent 9753 7865 IN IP4 192.168.47.119 s=SIP Call c=IN IP4 192.168.47.119 t=0 0 m=audio 17916 RTP/AVP 8 c=IN IP4 192.168.47.119 a=rtpmap:8 PCMA/8000 --- (13 headers 8 lines)--- Found RTP audio format 8 Peer audio RTP is at port 192.168.47.119:17916 Found description format PCMA Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.47.119, port 5060 Transmitting (no NAT) to 192.168.47.119:5060: ACK sip:2222222222@192.168.47.119:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK5a28d65e From: "1111111111" ;tag=as1a8fe718 To: ;tag=DB6858D0-75E Contact: Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/gateway-6602 answered SIP/192.168.47.119-081a4d20 We're at 192.168.47.101 port 10568 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 192.168.47.119:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.47.119:5060 From: ;tag=DB685690-1170 To: ;tag=as4ec92cda Call-ID: BC98F3B6-D33311D9-85ED9E35-AB6E44EE@192.168.47.119 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 21708 21709 IN IP4 192.168.47.101 s=session c=IN IP4 192.168.47.101 t=0 0 m=audio 10568 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/192.168.47.119-081a4d20 and SIP/gateway-6602 pluto*CLI> pluto*CLI> pluto*CLI> Sending to 192.168.47.119 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.47.119:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.47.119:5060 From: ;tag=DB685690-1170 To: ;tag=as4ec92cda Call-ID: BC98F3B6-D33311D9-85ED9E35-AB6E44EE@192.168.47.119 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.47.119, port 5060 Reliably Transmitting (no NAT) to 192.168.47.119:5060: BYE sip:2222222222@192.168.47.119:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK6705e22f From: "1111111111" ;tag=as1a8fe718 To: ;tag=DB6858D0-75E Contact: Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- == Spawn extension (remote, 9999999, 7) exited non-zero on 'SIP/192.168.47.119-081a4d20' pluto*CLI> <-- SIP read from 192.168.47.119:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.47.101:5060;branch=z9hG4bK6705e22f From: "1111111111" ;tag=as1a8fe718 To: ;tag=DB6858D0-75E Date: Fri, 03 Jun 2005 06:58:57 GMT Call-ID: 0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE --- (9 headers 0 lines)--- Destroying call '0f6d59b15466e4cb7952ef9316c7e427@192.168.47.101' Destroying call 'BC98F3B6-D33311D9-85ED9E35-AB6E44EE@192.168.47.119'