MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> INVITE sip:92819002620@192.168.100.180:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK57402c8e48A81CF7 From: "Jeff" ;tag=7159852B-F6C83D1C To: CSeq: 1 INVITE Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 300 v=0 o=- 1239892098 1239892098 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 a=sendrecv m=audio 2266 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> --- (16 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.101.250 : 5060 (no NAT) Using INVITE request as basis request - 153790c8-d1ea0841-8085cda2@192.168.101.250 Found peer 'sip1688' for 'sip1688' from 192.168.101.250:5060 <--- Reliably Transmitting (no NAT) to 192.168.101.250:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK57402c8e48A81CF7;received=192.168.101.250 From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as06d9ee90 Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c1d2efa" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '153790c8-d1ea0841-8085cda2@192.168.101.250' in 6400 ms (Method: INVITE) MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> ACK sip:92819002620@192.168.100.180:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK57402c8e48A81CF7 From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as06d9ee90 CSeq: 1 ACK Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> INVITE sip:92819002620@192.168.100.180:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK2a67fdb5A616D936 From: "Jeff" ;tag=7159852B-F6C83D1C To: CSeq: 2 INVITE Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="sip1688", realm="asterisk", nonce="1c1d2efa", uri="sip:92819002620@192.168.100.180:5060;user=phone", response="2405618eed4488cc3dcabd58c47d4a63", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 300 v=0 o=- 1239892098 1239892098 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 a=sendrecv m=audio 2266 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> --- (17 headers 13 lines) --- Sending to 192.168.101.250 : 5060 (no NAT) Using INVITE request as basis request - 153790c8-d1ea0841-8085cda2@192.168.101.250 Found peer 'sip1688' for 'sip1688' from 192.168.101.250:5060 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.101.250:2266 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.101.250:2266 Looking for 92819002620 in outgoing (domain 192.168.100.180) list_route: hop: <--- Transmitting (no NAT) to 192.168.101.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK2a67fdb5A616D936;received=192.168.101.250 From: "Jeff" ;tag=7159852B-F6C83D1C To: Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [92819002620@outgoing:1] Gosub("SIP/sip1688-09b86918", "dialout-callmanager,s,1(92819002620)") in new stack -- Executing [s@dialout-callmanager:1] ChanIsAvail("SIP/sip1688-09b86918", "SIP/callman01&SIP/callman02") in new stack == Using SIP RTP CoS mark 5 Scheduling destruction of SIP dialog '7972646e55b4cc952abaaafb2f788247@192.168.100.180' in 6400 ms (Method: INVITE) -- Executing [s@dialout-callmanager:2] Set("SIP/sip1688-09b86918", "AVAILCHAN=SIP/callman01") in new stack -- Executing [s@dialout-callmanager:3] NoOp("SIP/sip1688-09b86918", "92819002620 s") in new stack -- Executing [s@dialout-callmanager:4] Dial("SIP/sip1688-09b86918", "SIP/callman01/92819002620") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.100.180 port 17966 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 192.168.200.10:5060: INVITE sip:92819002620@192.168.200.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: Contact: Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Thu, 16 Apr 2009 14:28:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 295 v=0 o=root 788508858 788508858 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.180 t=0 0 m=audio 17966 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 callman01/92819002620 MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:32 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (9 headers 0 lines) --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:32 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 231 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.200.10 s=SIP Call c=IN IP4 192.168.200.10 t=0 0 m=audio 26822 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.200.10:26822 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), 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 192.168.200.10:26822 -- SIP/callman01-09b91890 is making progress passing it to SIP/sip1688-09b86918 Audio is at 192.168.100.180 port 13846 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.101.250:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK2a67fdb5A616D936;received=192.168.101.250 From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as2b64dd6d Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 295 v=0 o=root 327537650 327537650 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.180 t=0 0 m=audio 13846 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 <------------> MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:32 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 231 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.200.10 s=SIP Call c=IN IP4 192.168.200.10 t=0 0 m=audio 26822 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- -- SIP/callman01-09b91890 is making progress passing it to SIP/sip1688-09b86918 Really destroying SIP dialog '7972646e55b4cc952abaaafb2f788247@192.168.100.180' Method: INVITE MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:32 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 231 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.200.10 s=SIP Call c=IN IP4 192.168.200.10 t=0 0 m=audio 26822 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.10, port 5060 Transmitting (no NAT) to 192.168.200.10:5060: ACK sip:92819002620@192.168.200.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK372c28a2;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Contact: Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- -- SIP/callman01-09b91890 answered SIP/sip1688-09b86918 Audio is at 192.168.100.180 port 13846 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP MJLM-Hou-AsteriVOX-00*CLI> <--- Reliably Transmitting (no NAT) to 192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK2a67fdb5A616D936;received=192.168.101.250 From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as2b64dd6d Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 295 v=0 o=root 327537650 327537651 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.180 t=0 0 m=audio 13846 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 <------------> -- Native bridging SIP/sip1688-09b86918 and SIP/callman01-09b91890 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.10, port 5060 Audio is at 192.168.100.180 port 17966 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.200.10:5060: INVITE sip:92819002620@192.168.200.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Contact: Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 270 v=0 o=root 788508858 788508859 IN IP4 192.168.101.250 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.101.250 t=0 0 m=audio 2266 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 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:40 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 103 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=yes;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:40 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 231 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.200.10 s=SIP Call c=IN IP4 192.168.200.10 t=0 0 m=audio 26822 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.10, port 5060 Transmitting (no NAT) to 192.168.200.10:5060: ACK sip:92819002620@192.168.200.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK5502d7d5;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Contact: Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 ---M-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> ACK sip:92819002620@192.168.100.180 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bK8143a9a97A199F4A From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as2b64dd6d CSeq: 2 ACK Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.101.250, port 5060 Audio is at 192.168.100.180 port 13846 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 192.168.101.250:5060: INVITE sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK73755d1f;rport Max-Forwards: 70 From: ;tag=as2b64dd6d To: "Jeff" ;tag=7159852B-F6C83D1C Contact: Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 293 v=0 o=root 327537650 327537652 IN IP4 192.168.200.10 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.200.10 t=0 0 m=audio 26822 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 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK73755d1f;rport From: ;tag=as2b64dd6d To: "Jeff" ;tag=7159852B-F6C83D1C CSeq: 102 INVITE Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Type: application/sdp Content-Length: 205 v=0 o=- 1239892098 1239892099 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 m=audio 2266 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0I> Found RTP audio format 101 Peer audio RTP is at port 192.168.101.250:2266 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), 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 192.168.101.250:2266 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.101.250, port 5060 Transmitting (no NAT) to 192.168.101.250:5060: ACK sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK7c4050a7;rport Max-Forwards: 70 From: ;tag=as2b64dd6d To: "Jeff" ;tag=7159852B-F6C83D1C Contact: Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> BYE sip:92819002620@192.168.100.180 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bKfa4dd113BACCDD44 From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as2b64dd6d CSeq: 3 BYE Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Authorization: Digest username="sip1688", realm="asterisk", nonce="1c1d2efa", uri="sip:92819002620@192.168.100.180:5060;user=phone", response="ccbb230eca0bc97648478f092fc9516a", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.101.250 : 5060 (no NAT) MJLM-Hou-AsteriVOX-00*CLI> <--- Transmitting (no NAT) to 192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bKfa4dd113BACCDD44;received=192.168.101.250 From: "Jeff" ;tag=7159852B-F6C83D1C To: ;tag=as2b64dd6d Call-ID: 153790c8-d1ea0841-8085cda2@192.168.101.250 CSeq: 3 BYE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 MJLM-Hou-AsteriVOX-00*CLI> <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.10, port 5060 Audio is at 192.168.100.180 port 17966 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.200.10:5060: INVITE sip:92819002620@192.168.200.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Contact: Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 271 v=0 o=root 788508858 788508860 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.180 t=0 0 m=audio 17966 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 --- Scheduling destruction of SIP dialog '607825a258ff429b39afc10d67950a3a@192.168.100.180' in 6400 ms (Method: INVITE) == Spawn extension (dialout-callmanager, s, 4) exited non-zero on 'SIP/sip1688-09b86918' MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:47 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 104 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '153790c8-d1ea0841-8085cda2@192.168.101.250' Method: BYE MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1f6165db;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:47 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 231 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.200.10 s=SIP Call c=IN IP4 192.168.200.10 t=0 0 m=audio 26822 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.10, port 5060 Transmitting (no NAT) to 192.168.200.10:5060: ACK sip:92819002620@192.168.200.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK78e26022;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Contact: Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.10, port 5060 Reliably Transmitting (no NAT) to 192.168.200.10:5060: BYE sip:92819002620@192.168.200.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK73d650a0;rport Max-Forwards: 70 From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 CSeq: 105 BYE User-Agent: Asterisk PBX 1.6.1.0-rc4 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '607825a258ff429b39afc10d67950a3a@192.168.100.180' in 6400 ms (Method: INVITE) MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK73d650a0;rport From: "Jeff Phelps" ;tag=as104cfc80 To: ;tag=17010960 Date: Fri, 17 Apr 2009 02:34:47 GMT Call-ID: 607825a258ff429b39afc10d67950a3a@192.168.100.180 Content-Length: 0 CSeq: 105 BYE <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '607825a258ff429b39afc10d67950a3a@192.168.100.180' Method: INVITE MJLM-Hou-AsteriVOX-00*CLI> sip set debug off SIP Debugging DisabledCLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI>