Sending to 66.250.69.11 : 5060 (non-NAT) Found peer 'asterlink-switch-01' Found RTP audio format 10 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 111 Found RTP audio format 5 Found RTP audio format 7 Found RTP audio format 18 Found RTP audio format 110 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 66.250.69.11:23426 Found description format L16 Found description format G723 Found description format GSM Found description format PCMU Found description format PCMA Found description format G726-32 Found description format DVI4 Found description format LPC Found description format G729 Found description format speex Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 8006170141 in incoming (domain host41.com) list_route: hop: Transmitting (no NAT) to 66.250.69.11:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK16f07be1;received=66.250.69.11 From: "4072930335" ;tag=as4536f66d To: Call-ID: 798134131272577c3f910c4425ee6cb3@66.250.69.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 10 18:21:07 NOTICE[29960]: chan_sip.c:5342 sip_reregister: -- Re-registration for 120983@switch-01.asterlink.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 66.250.69.11:5060: REGISTER sip:switch-01.asterlink.com SIP/2.0 Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK1ec289a1;rport From: ;tag=as61f0ccdb To: Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13 CSeq: 1438 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="120983", realm="asterisk", algorithm=MD5, uri="sip:switch-01.asterlink.com", nonce="4c9531cd", response="5f70a8a142e8c4d33a17fd1d58448b0c", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- aries*CLI> <-- SIP read from 66.250.69.11:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK1ec289a1 From: ;tag=as61f0ccdb To: Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13 CSeq: 1438 REGISTER User-Agent: asterlink/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- aries*CLI> <-- SIP read from 66.250.69.11:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK1ec289a1 From: ;tag=as61f0ccdb To: ;tag=as02ea1de8 Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13 CSeq: 1438 REGISTER User-Agent: asterlink/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="036614d7" Content-Length: 0 --- (11 headers 0 lines)--- Responding to challenge, registration to domain/host name switch-01.asterlink.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 66.250.69.11:5060: REGISTER sip:switch-01.asterlink.com SIP/2.0 Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK7c6ea1e6;rport From: ;tag=as44a8d573 To: Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13 CSeq: 1439 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="120983", realm="asterisk", algorithm=MD5, uri="sip:switch-01.asterlink.com", nonce="036614d7", response="8fdbbdbd65d16cf1804a429c8a670cb5", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- aries*CLI> <-- SIP read from 66.250.69.11:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK7c6ea1e6 From: ;tag=as44a8d573 To: Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13 CSeq: 1439 REGISTER User-Agent: asterlink/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- aries*CLI> <-- SIP read from 66.250.69.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK7c6ea1e6 From: ;tag=as44a8d573 To: ;tag=as02ea1de8 Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13 CSeq: 1439 REGISTER User-Agent: asterlink/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 120 Contact: ;expires=120 Date: Mon, 10 Jul 2006 22:19:46 GMT Content-Length: 0 aries*CLI> --- (12 headers 0 lines)--- Scheduling destruction of call '64d8dd26442a40524a79fe062763ab50@72.232.182.13' in 32000 ms Jul 10 18:21:07 NOTICE[29960]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for switch-01.asterlink.com is 120 sec (Scheduling reregistration in 105 s) We're at 72.232.182.13 port 10610 Adding codec 0x2 (gsm) to SDP 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 66.250.69.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK16f07be1;received=66.250.69.11 From: "4072930335" ;tag=as4536f66d To: ;tag=as13bd9c93 Call-ID: 798134131272577c3f910c4425ee6cb3@66.250.69.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 29951 29951 IN IP4 72.232.182.13 s=session c=IN IP4 72.232.182.13 t=0 0 m=audio 10610 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 - - - - --- aries*CLI> <-- SIP read from 66.250.69.11:5060: ACK sip:8006170141@72.232.182.13 SIP/2.0 Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK1ef7d78c From: "4072930335" ;tag=as4536f66d To: ;tag=as13bd9c93 Contact: Call-ID: 798134131272577c3f910c4425ee6cb3@66.250.69.11 CSeq: 102 ACK User-Agent: asterlink/1.2 Content-Length: 0 --- (9 headers 0 lines)--- We're at 72.232.182.13 port 18688 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 72.40.10.150:5060: INVITE sip:200@72.40.10.150:5060 SIP/2.0 Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK637df9c1;rport From: "4072930335" ;tag=as1df7cef9 To: Contact: Call-ID: 290b3a4b583911136c9a4a224278040b@72.232.182.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 10 Jul 2006 22:21:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 29951 29951 IN IP4 72.232.182.13 s=session c=IN IP4 72.232.182.13 t=0 0 m=audio 18688 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 10 18:21:19 WARNING[21690]: res_musiconhold.c:896 local_ast_moh_start: No class: default aries*CLI> <-- SIP read from 72.40.10.150:5060: SIP/2.0 404 Not Found To: ;tag=656eae54128438bbi0 From: "4072930335" ;tag=as1df7cef9 Call-ID: 290b3a4b583911136c9a4a224278040b@72.232.182.13 CSeq: 102 INVITE Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK637df9c1 Server: Linksys/PAP2-3.1.3(LS) Content-Length: 0 --- (8 headers 0 lines)--- Transmitting (no NAT) to 72.40.10.150:5060: ACK sip:200@72.40.10.150:5060 SIP/2.0 Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK637df9c1;rport From: "4072930335" ;tag=as1df7cef9 To: ;tag=656eae54128438bbi0 Contact: Call-ID: 290b3a4b583911136c9a4a224278040b@72.232.182.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '290b3a4b583911136c9a4a224278040b@72.232.182.13' aries*CLI>