LOCAL SIDE local*CLI> sip debug ip remote SIP Debugging Enabled for IP: 66.0.0.1 Sip read: INVITE sip:18005555555@66.0.2.1 SIP/2.0 Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK3b01f2ba From: "Unavailable" ;tag=as2fabd720 To: Contact: Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 08 Jul 2005 19:15:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 366 v=0 o=root 8860 8860 IN IP4 66.0.0.1 s=session c=IN IP4 66.0.0.1 t=0 0 m=audio 10180 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 12 headers, 16 lines Using latest request as basis request Sending to 66.0.0.1 : 5060 (non-NAT) Found no matching peer or user for '66.0.0.1:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 66.0.0.1:10180 Found description format PCMU Found description format PCMA Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51f (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Looking for 18005555555 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK3b01f2ba From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 66.0.0.1:5060 We're at 66.0.2.1 port 15638 Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK3b01f2ba From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 21728 21728 IN IP4 66.0.2.1 s=session c=IN IP4 66.0.2.1 t=0 0 m=audio 15638 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 - - - - to 66.0.0.1:5060 local*CLI> Sip read: ACK sip:18005555555@66.0.2.1 SIP/2.0 Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK650b7da1 From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Contact: Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 9 headers, 0 lines local*CLI> Sip read: BYE sip:18005555555@66.0.2.1 SIP/2.0 Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK2a732dea From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Contact: Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 9 headers, 0 lines Sending to 66.0.0.1 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK2a732dea From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 66.0.0.1:5060 Destroying call '6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1' REMOTE SIDE remote*CLI> sip debug ip local SIP Debugging Enabled for IP: 66.0.2.1 We're at 66.0.0.1 port 10180 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 16 lines Reliably Transmitting: INVITE sip:18005555555@66.0.2.1 SIP/2.0 Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK3b01f2ba From: "Unavailable" ;tag=as2fabd720 To: Contact: Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 08 Jul 2005 19:15:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 366 v=0 o=root 8860 8860 IN IP4 66.0.0.1 s=session c=IN IP4 66.0.0.1 t=0 0 m=audio 10180 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 66.0.2.1:5060 remote*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK3b01f2ba From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 10 headers, 0 lines remote*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK3b01f2ba From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 21728 21728 IN IP4 66.0.2.1 s=session c=IN IP4 66.0.2.1 t=0 0 m=audio 15638 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 - - - - 11 headers, 12 lines Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 66.0.2.1:15638 Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x51f (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 66.0.2.1, port 5060 Transmitting: ACK sip:18005555555@66.0.2.1 SIP/2.0 Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK650b7da1 From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Contact: Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 66.0.2.1:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 66.0.2.1, port 5060 Reliably Transmitting: BYE sip:18005555555@66.0.2.1 SIP/2.0 Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK2a732dea From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Contact: Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 66.0.2.1:5060 remote*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.0.0.1:5060;branch=z9hG4bK2a732dea From: "Unavailable" ;tag=as2fabd720 To: ;tag=as25139def Call-ID: 6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 10 headers, 0 lines Destroying call '6783d7ef049fd23a592f0fd57e3a2ff2@66.0.0.1' remote*CLI> sip no debug SIP Debugging Disabled