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November 8, 2010 01:42
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| mbp-linux*CLI> sip set debug on | |
| SIP Debugging enabled | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:192.168.1.64:43533 ---> | |
| <-------------> | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:192.168.1.64:59247 --->time | |
| <-------------> | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| INVITE sip:5594030000@192.168.1.65 SIP/2.0 | |
| Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKDpU80pZmpgm4c | |
| Max-Forwards: 49 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=cpS1crj11HatQ | |
| To: <sip:5594030000@192.168.1.65> | |
| Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 INVITE | |
| Contact: <sip:mod_sofia@216.82.225.24:5060> | |
| User-Agent: FreePBX Trunking | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Privacy: none | |
| Content-Type: application/sdp | |
| Content-Disposition: session | |
| Content-Length: 270 | |
| P-Asserted-Identity: "+15596960400" <sip:5596960400@216.82.225.24> | |
| v=0 | |
| o=Sonus_UAC 7878 27307 IN IP4 192.168.47.68 | |
| s=SIP Media Capabilities | |
| c=IN IP4 67.231.4.99 | |
| t=0 0 | |
| m=audio 14988 RTP/AVP 0 18 101 | |
| a=rtpmap:0 PCMU/8000 | |
| a=rtpmap:18 G729/8000 | |
| a=fmtp:18 annexb=no | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-15 | |
| a=maxptime:20 | |
| <-------------> | |
| --- (17 headers 12 lines) --- | |
| Sending to 216.82.225.24 : 5060 (NAT) | |
| Using INVITE request as basis request - 292f1216-657b-122e-dfbf-0015c5eaaddb | |
| Found peer 'ext-sip-account' for '5596960400' from 216.82.225.24:5060 | |
| Found RTP audio format 0 | |
| Found RTP audio format 18 | |
| Found RTP audio format 101 | |
| Found audio description format PCMU for ID 0 | |
| Found audio description format G729 for ID 18 | |
| Found audio description format telephone-event for ID 101 | |
| Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=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 67.231.4.99:14988 | |
| Looking for 5594030000 in default (domain 192.168.1.65) | |
| list_route: hop: <sip:mod_sofia@216.82.225.24:5060> | |
| <--- Transmitting (no NAT) to 216.82.225.24:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKDpU80pZmpgm4c;received=216.82.225.24;rport=5060 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=cpS1crj11HatQ | |
| To: <sip:5594030000@192.168.1.65> | |
| Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 INVITE | |
| Server: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Require: timer | |
| Session-Expires: 1800;refresher=uas | |
| Contact: <sip:5594030000@99.28.157.10> | |
| Content-Length: 0 | |
| <------------> | |
| Audio is at 99.28.157.10 port 19242 | |
| Adding codec 0x4 (ulaw) to SDP | |
| Adding non-codec 0x1 (telephone-event) to SDP | |
| Reliably Transmitting (no NAT) to 216.82.225.24:5060: | |
| INVITE sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK1a40b322;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 102 INVITE | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Date: Mon, 08 Nov 2010 01:34:24 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Content-Type: application/sdp | |
| Content-Length: 270 | |
| v=0 | |
| o=root 261348102 261348102 IN IP4 99.28.157.10 | |
| s=Asterisk PBX 1.6.2.7-1ubuntu1 | |
| c=IN IP4 99.28.157.10 | |
| t=0 0 | |
| m=audio 19242 RTP/AVP 0 101 | |
| a=rtpmap:0 PCMU/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=silenceSupp:off - - - - | |
| a=ptime:20 | |
| a=sendrecv | |
| --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1a40b322;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 102 INVITE | |
| User-Agent: FreePBX Trunking | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 407 Proxy Authentication Required | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1a40b322;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=DZjteK34yt0cK | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 102 INVITE | |
| User-Agent: FreePBX Trunking | |
| Accept: application/sdp | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Proxy-Authenticate: Digest realm="trunk1.freepbx.com", nonce="51d6a718-ead8-11df-8944-edffed59eac9", algorithm=MD5, qop="auth" | |
| Content-Length: 0 | |
| <-------------> | |
| --- (13 headers 0 lines) --- | |
| Transmitting (no NAT) to 216.82.225.24:5060: | |
| ACK sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK1a40b322;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=DZjteK34yt0cK | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Content-Length: 0 | |
| --- | |
| Audio is at 99.28.157.10 port 19242 | |
| Adding codec 0x4 (ulaw) to SDP | |
| dding non-codec 0x1 (telephone-event) to SDP | |
| Reliably Transmitting (no NAT) to 216.82.225.24:5060: | |
| INVITE sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK31ff4f96;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 103 INVITE | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Proxy-Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:5594030000@trunk1.freepbx.com", nonce="51d6a718-ead8-11df-8944-edffed59eac9", response="6a2d195a824230445425db4cbd11ba5a", qop=auth, cnonce="5b001aea", nc=00000001 | |
| Date: Mon, 08 Nov 2010 01:34:25 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Content-Type: application/sdp | |
| Content-Length: 270 | |
| v=0 | |
| o=root 261348102 261348103 IN IP4 99.28.157.10 | |
| s=Asterisk PBX 1.6.2.7-1ubuntu1 | |
| c=IN IP4 99.28.157.10 | |
| t=0 0 | |
| m=audio 19242 RTP/AVP 0 101 | |
| a=rtpmap:0 PCMU/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=silenceSupp:off - - - - | |
| a=ptime:20 | |
| a=sendrecv | |
| --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK31ff4f96;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 103 INVITE | |
| User-Agent: FreePBX Trunking | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| INVITE sip:5594030000@192.168.1.65 SIP/2.0 | |
| Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKeZm12HgrKSaQr | |
| Max-Forwards: 69 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=FH5Bj94BScDja | |
| To: <sip:5594030000@192.168.1.65> | |
| Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 INVITE | |
| Contact: <sip:mod_sofia@216.82.225.24:5060> | |
| User-Agent: FreePBX Trunking | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Content-Type: application/sdp | |
| Content-Disposition: session | |
| Content-Length: 258 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@216.82.225.24>;party=calling;screen=no;privacy=off | |
| v=0 | |
| o=root 261348102 261348103 IN IP4 99.28.157.10 | |
| s=Asterisk PBX 1.6.2.7-1ubuntu1 | |
| c=IN IP4 99.28.157.10 | |
| t=0 0 | |
| m=audio 19242 RTP/AVP 0 101 | |
| a=rtpmap:0 PCMU/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=silenceSupp:off - - - - | |
| a=ptime:20 | |
| <-------------> | |
| --- (16 headers 11 lines) --- | |
| Sending to 216.82.225.24 : 5060 (NAT) | |
| Using INVITE request as basis request - 2941d1a7-657b-122e-dfbf-0015c5eaaddb | |
| Found peer 'ext-sip-account' for '5596960400' from 216.82.225.24:5060 | |
| Found RTP audio format 0 | |
| Found RTP audio format 101 | |
| Found audio description format PCMU for ID 0 | |
| Found audio description format telephone-event for ID 101 | |
| Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 99.28.157.10:19242 | |
| Looking for 5594030000 in default (domain 192.168.1.65) | |
| list_route: hop: <sip:mod_sofia@216.82.225.24:5060> | |
| <--- Transmitting (no NAT) to 216.82.225.24:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKeZm12HgrKSaQr;received=216.82.225.24;rport=5060 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=FH5Bj94BScDja | |
| To: <sip:5594030000@192.168.1.65> | |
| Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 INVITE | |
| Server: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Require: timer | |
| Session-Expires: 1800;refresher=uas | |
| Contact: <sip:5594030000@99.28.157.10> | |
| Content-Length: 0 | |
| <------------> | |
| Audio is at 99.28.157.10 port 16054 | |
| Adding codec 0x4 (ulaw) to SDP | |
| Adding non-codec 0x1 (telephone-event) to SDP | |
| Reliably Transmitting (no NAT) to 216.82.225.24:5060: | |
| INVITE sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK34442773;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 102 INVITE | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Date: Mon, 08 Nov 2010 01:34:25 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Content-Type: application/sdp | |
| Content-Length: 272 | |
| v=0 | |
| o=root 1191172620 1191172620 IN IP4 99.28.157.10 | |
| s=Asterisk PBX 1.6.2.7-1ubuntu1 | |
| c=IN IP4 99.28.157.10 | |
| t=0 0 | |
| m=audio 16054 RTP/AVP 0 101 | |
| a=rtpmap:0 PCMU/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=silenceSupp:off - - - - | |
| a=ptime:20 | |
| a=sendrecv | |
| --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK34442773;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 102 INVITE | |
| User-Agent: FreePBX Trunking | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 407 Proxy Authentication Required | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK34442773;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=gty4K4NFpN34N | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 102 INVITE | |
| User-Agent: FreePBX Trunking | |
| Accept: application/sdp | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Proxy-Authenticate: Digest realm="trunk1.freepbx.com", nonce="51e923d4-ead8-11df-8944-edffed59eac9", algorithm=MD5, qop="auth" | |
| Content-Length: 0 | |
| <-------------> | |
| --- (13 headers 0 lines) --- | |
| Transmitting (no NAT) to 216.82.225.24:5060: | |
| ACK sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK34442773;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=gty4K4NFpN34N | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Content-Length: 0 | |
| --- | |
| Audio is at 99.28.157.10 port 16054 | |
| Adding codec 0x4 (ulaw) to SDP | |
| dding non-codec 0x1 (telephone-event) to SDP | |
| Reliably Transmitting (no NAT) to 216.82.225.24:5060: | |
| INVITE sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK6eda181a;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 103 INVITE | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Proxy-Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:5594030000@trunk1.freepbx.com", nonce="51e923d4-ead8-11df-8944-edffed59eac9", response="64d622927159ddfe8fe699044db4fba2", qop=auth, cnonce="049158b8", nc=00000001 | |
| Date: Mon, 08 Nov 2010 01:34:25 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Content-Type: application/sdp | |
| Content-Length: 272 | |
| v=0 | |
| o=root 1191172620 1191172621 IN IP4 99.28.157.10 | |
| s=Asterisk PBX 1.6.2.7-1ubuntu1 | |
| c=IN IP4 99.28.157.10 | |
| t=0 0 | |
| m=audio 16054 RTP/AVP 0 101 | |
| a=rtpmap:0 PCMU/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=silenceSupp:off - - - - | |
| a=ptime:20 | |
| a=sendrecv | |
| --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK6eda181a;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com> | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 103 INVITE | |
| User-Agent: FreePBX Trunking | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 480 Temporarily Unavailable | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK6eda181a;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=KNaFSN8SDg6vr | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 103 INVITE | |
| User-Agent: FreePBX Trunking | |
| Accept: application/sdp | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Reason: Q.850;cause=16;text="NORMAL_CLEARING" | |
| Content-Length: 0 | |
| <-------------> | |
| --- (13 headers 0 lines) --- | |
| Transmitting (no NAT) to 216.82.225.24:5060: | |
| ACK sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK6eda181a;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as5c4b531a | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=KNaFSN8SDg6vr | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 33d374a1082e01a520b826892acf92f6@99.28.157.10 | |
| CSeq: 103 ACK | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Content-Length: 0 | |
| --- | |
| <--- Reliably Transmitting (no NAT) to 216.82.225.24:5060 ---> | |
| SIP/2.0 503 Service Unavailable | |
| Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKeZm12HgrKSaQr;received=216.82.225.24;rport=5060 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=FH5Bj94BScDja | |
| To: <sip:5594030000@192.168.1.65>;tag=as08867e89 | |
| Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 INVITE | |
| Server: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Require: timer | |
| Session-Expires: 1800;refresher=uas | |
| Content-Length: 0 | |
| X-Asterisk-HangupCause: User alerting, no answer | |
| X-Asterisk-HangupCauseCode: 19 | |
| <------------> | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| ACK sip:5594030000@192.168.1.65 SIP/2.0 | |
| Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKeZm12HgrKSaQr | |
| Max-Forwards: 69 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=FH5Bj94BScDja | |
| To: <sip:5594030000@192.168.1.65:5060>;tag=as08867e89 | |
| Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 ACK | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| Really destroying SIP dialog '33d374a1082e01a520b826892acf92f6@99.28.157.10' Method: INVITE | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 486 Busy Here | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK31ff4f96;rport=5060 | |
| From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=e8BKgem8U3pZe | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 103 INVITE | |
| Contact: <sip:mod_sofia@216.82.225.24:5060;transport=udp> | |
| User-Agent: FreePBX Trunking | |
| Accept: application/sdp | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Content-Length: 0 | |
| <-------------> | |
| --- (13 headers 0 lines) --- | |
| Transmitting (no NAT) to 216.82.225.24:5060: | |
| ACK sip:5594030000@trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK31ff4f96;rport | |
| Max-Forwards: 70 | |
| From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as662ebece | |
| To: <sip:5594030000@trunk1.freepbx.com>;tag=e8BKgem8U3pZe | |
| Contact: <sip:100ce039@99.28.157.10> | |
| Call-ID: 7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10 | |
| CSeq: 103 ACK | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no | |
| Content-Length: 0 | |
| --- | |
| <--- Reliably Transmitting (no NAT) to 216.82.225.24:5060 ---> | |
| SIP/2.0 486 Busy Here | |
| Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKDpU80pZmpgm4c;received=216.82.225.24;rport=5060 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=cpS1crj11HatQ | |
| To: <sip:5594030000@192.168.1.65>;tag=as75eb340a | |
| Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 INVITE | |
| Server: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Require: timer | |
| Session-Expires: 1800;refresher=uas | |
| Content-Length: 0 | |
| X-Asterisk-HangupCause: User busy | |
| X-Asterisk-HangupCauseCode: 17 | |
| <------------> | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| ACK sip:5594030000@192.168.1.65 SIP/2.0 | |
| Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKDpU80pZmpgm4c | |
| Max-Forwards: 49 | |
| From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=cpS1crj11HatQ | |
| To: <sip:5594030000@192.168.1.65:5060>;tag=as75eb340a | |
| Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb | |
| CSeq: 4253968 ACK | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| Really destroying SIP dialog '7e4bf7cc5a42a31f15d9092274ea7571@99.28.157.10' Method: INVITE | |
| Really destroying SIP dialog '02ed5599528d004b5160d976293943e5@192.168.1.65' Method: REGISTER | |
| Reliably Transmitting (no NAT) to 216.82.225.24:5060: | |
| OPTIONS sip:trunk1.freepbx.com SIP/2.0 | |
| Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK52cfa45e;rport | |
| Max-Forwards: 70 | |
| From: "asterisk" <sip:asterisk@99.28.157.10>;tag=as1880ca6b | |
| To: <sip:trunk1.freepbx.com> | |
| Contact: <sip:asterisk@99.28.157.10> | |
| Call-ID: 42de250f5c20143e5e95a01f24e35235@99.28.157.10 | |
| CSeq: 102 OPTIONS | |
| User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1 | |
| Date: Mon, 08 Nov 2010 01:34:42 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO | |
| Supported: replaces, timer | |
| Content-Length: 0 | |
| --- | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:216.82.225.24:5060 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK52cfa45e;rport=5060 | |
| From: "asterisk" <sip:asterisk@192.168.1.65>;tag=as1880ca6b | |
| To: <sip:trunk1.freepbx.com>;tag=g95Z7yBcBa9je | |
| Call-ID: 42de250f5c20143e5e95a01f24e35235@99.28.157.10 | |
| CSeq: 102 OPTIONS | |
| Contact: <sip:216.82.225.24> | |
| User-Agent: FreePBX Trunking | |
| Accept: application/sdp | |
| Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH | |
| Supported: timer, precondition, path, replaces | |
| Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer | |
| Content-Length: 0 | |
| <-------------> | |
| --- (13 headers 0 lines) --- | |
| Really destroying SIP dialog '42de250f5c20143e5e95a01f24e35235@99.28.157.10' Method: OPTIONS | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:192.168.1.64:43533 ---> | |
| <-------------> | |
| mbp-linux*CLI> | |
| <--- SIP read from UDP:192.168.1.64:59247 ---> | |
| <-------------> |
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