Asterisk CVS-HEAD-06/25/04-17:24:00, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-06/25/04-17:24:00 currently running on ast1000 (pid = 26747) ast1000*CLI> -- Remote UNIX connection ast1000*CLI> sip debug ast1000*CLI> SIP Debugging Enabled ast1000*CLI> Sip read: INVITE sip:*393613@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-f2a340d1;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 101 INVITE Max-Forwards: 70 Contact: Kai-Uwe Jensen Expires: 240 User-Agent: Sipura/SPA2000-2.0.9(d) Content-Length: 407 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 20457818 20457818 IN IP4 192.168.254.251 s=- c=IN IP4 192.168.254.251 t=0 0 m=audio 9070 RTP/AVP 2 0 8 18 96 97 98 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 14 headers, 18 lines Using latest request as basis request Sending to 192.168.254.251 : 5061 (non-NAT) Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer RTP is at port 192.168.254.251:0 Found description format G726-32 Found description format PCMU Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x16(GSM|ULAW|G726), peer - audio=0x51c(ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x14(ULAW|G726) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-f2a340d1;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: ;tag=as5b86c59e Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="cojensen.net", nonce="4c474018" Content-Length: 0 to 192.168.254.251:5061 Scheduling destruction of call 'c5d56060-3a4076f9@192.168.254.251' in 15000 ms Found user '2001' ast1000*CLI> Sip read: ACK sip:*393613@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-f2a340d1;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: ;tag=as5b86c59e Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 101 ACK Max-Forwards: 70 Contact: Kai-Uwe Jensen User-Agent: Sipura/SPA2000-2.0.9(d) Content-Length: 0 10 headers, 0 lines ast1000*CLI> Sip read: INVITE sip:*393613@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-39a013b5;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="cojensen.net",nonce="4c474018",uri="sip:*393613@192.168.254.250",algorithm=MD5,response="ff66ea319bd3a89ca637e9324dccdf56" Contact: Kai-Uwe Jensen Expires: 240 User-Agent: Sipura/SPA2000-2.0.9(d) Content-Length: 407 Content-Type: application/sdp v=0 o=- 20457818 20457818 IN IP4 192.168.254.251 s=- c=IN IP4 192.168.254.251 t=0 0 m=audio 9070 RTP/AVP 2 0 8 18 96 97 98 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 13 headers, 18 lines Using latest request as basis request Sending to 192.168.254.251 : 5061 (non-NAT) Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer RTP is at port 192.168.254.251:0 Found description format G726-32 Found description format PCMU Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x16(GSM|ULAW|G726), peer - audio=0x51c(ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x14(ULAW|G726) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '2001' Looking for *393613 in home list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-39a013b5;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: ;tag=as26817e03 Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.254.251:5061 -- Executing Playback("SIP/2001-7bf1", "tr-fwd") in new stack ast1000*CLI> We're at 192.168.254.250 port 9028 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x10(G726) Answering with preferred capability 0x2(GSM) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-39a013b5;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: ;tag=as26817e03 Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 26747 26747 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9028 RTP/AVP 0 2 3 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.254.251:5061 -- Playing 'tr-fwd' (language 'en') ast1000*CLI> Sip read: ACK sip:*393613@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.251:5061;branch=z9hG4bK-fc36d3ac;rport From: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a To: ;tag=as26817e03 Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="cojensen.net",nonce="4c474018",uri="sip:*393613@192.168.254.250",algorithm=MD5,response="034aad818cde27e36c5ba8fa2484ccdc" Contact: Kai-Uwe Jensen User-Agent: Sipura/SPA2000-2.0.9(d) Content-Length: 0 11 headers, 0 lines ast1000*CLI> -- Executing Macro("SIP/2001-7bf1", "dialfwd|613|60") in new stack -- Executing SetCallerID("SIP/2001-7bf1", "13607171717") in new stack -- Executing SetCIDName("SIP/2001-7bf1", "Kai-Uwe Jensen") in new stack -- Executing Dial("SIP/2001-7bf1", "SIP/613@fwd|60") in new stack We're at 67.165.241.16 port 9026 Answering/Requesting with root capability 16 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x2(GSM) Answering with non-codec capability 0x1(G723) 12 headers, 12 lines Reliably Transmitting: INVITE sip:613@fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK1b5cc0eb From: "Kai-Uwe Jensen" ;tag=as7323a684 To: Contact: Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 02 Jul 2004 01:43:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 267 v=0 o=root 26747 26747 IN IP4 67.165.241.16 s=session c=IN IP4 67.165.241.16 t=0 0 m=audio 9026 RTP/AVP 2 0 3 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.246.69.223:5060 -- Called 613@fwd ast1000*CLI> Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK1b5cc0eb From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=ec5154884c15db89cef5049aa4f06328.b65c Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="fwd.pulver.com", nonce="40e4bed3860d14568dcd2214ad9f37d5568d7f9f" Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:613@fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK1b5cc0eb From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=ec5154884c15db89cef5049aa4f06328.b65c Contact: Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.246.69.223:5060 We're at 67.165.241.16 port 9026 Answering/Requesting with root capability 16 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x2(GSM) Answering with non-codec capability 0x1(G723) Reliably Transmitting: INVITE sip:613@fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK35163583 From: "Kai-Uwe Jensen" ;tag=as7323a684 To: Contact: Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="256222", realm="fwd.pulver.com", algorithm="MD5", uri="sip:613@fwd.pulver.com", nonce="40e4bed3860d14568dcd2214ad9f37d5568d7f9f", response="ff8edb1bf489b057b0117a6f0e3bda10", opaque="" Date: Fri, 02 Jul 2004 01:43:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 267 v=0 o=root 26747 26748 IN IP4 67.165.241.16 s=session c=IN IP4 67.165.241.16 t=0 0 m=audio 9026 RTP/AVP 2 0 3 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.246.69.223:5060 ast1000*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK35163583 From: "Kai-Uwe Jensen" ;tag=as7323a684 To: Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 103 INVITE Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines ast1000*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK35163583 From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=as2e2179d6 Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 10 headers, 0 lines -- SIP/fwd-b1d1 is ringing ast1000*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK35163583 Record-Route: From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=as2e2179d6 Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 216 v=0 o ast1000*CLI> =root 26968 26968 IN IP4 69.90.168.13 s=session c=IN IP4 69.90.168.13 t=0 0 m=audio 19982 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 12 headers, 10 lines Found RTP audio format 0 Found RTP audio format 101 Peer RTP is at port 69.90.168.13:0 Found description format PCMU Found description format telephone-event Capabilities: us - 0x16(GSM|ULAW|G726), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: list_route: hop: -- SIP/fwd-b1d1 answered SIP/2001-7bf1 set_destination: Parsing for address/port to send to ast1000*CLI> set_destination: set destination to 192.246.69.223, port 5060 ast1000*CLI> Transmitting: ACK sip:613@69.90.168.13:5028 SIP/2.0 Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK136f451b Route: From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=as2e2179d6 Contact: Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.246.69.223:5060 ast1000*CLI> -- Attempting native bridge of SIP/2001-7bf1 and SIP/fwd-b1d1 -- Attempting native bridge of SIP/2001-7bf1 and SIP/fwd-b1d1 ast1000*CLI> Jul 1 19:43:10 WARNING[-1234187344]: chan_sip.c:1800 sip_write: Asked to transmit frame type 64, while native formats is 16 (read/write = 4/16) set_destination: Parsing for address/port to send to set_destination: set destination to 192.246.69.223, port 5060 Reliably Transmitting: BYE sip:613@69.90.168.13:5028 SIP/2.0 Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK73b58f8c Route: From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=as2e2179d6 Contact: Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 104 BYE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="256222", realm="fwd.pulver.com", algorithm="MD5", uri="sip:613@69.90.168.13:5028", nonce="40e4bed3860d14568dcd2214ad9f37d5568d7f9f", response="6440d103d604faa1f99aece5eebcb64a", opaque="" Content-Length: 0 (no NAT) to 192.246.69.223:5060 == Spawn extension (macro-dialfwd, s, 3) exited non-zero on 'SIP/2001-7bf1' in macro 'dialfwd' == Spawn extension (home, *393613, 2) exited non-zero on 'SIP/2001-7bf1' ast1000*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.251, port 5061 Reliably Transmitting: BYE sip:2001@192.168.254.251:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK172cadb2;rport From: ;tag=as26817e03 To: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a Contact: Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.254.251:5061 ast1000*CLI> Sip read: SIP/2.0 200 OK To: Kai-Uwe Jensen ;tag=7c3c773ffd15e9a From: ;tag=as26817e03 Call-ID: c5d56060-3a4076f9@192.168.254.251 CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK172cadb2;rport=5060 Server: Sipura/SPA2000-2.0.9(d) Content-Length: 0 8 headers, 0 lines Message is BYE Destroying call 'c5d56060-3a4076f9@192.168.254.251' ast1000*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.165.241.16:5060;branch=z9hG4bK73b58f8c Record-Route: From: "Kai-Uwe Jensen" ;tag=as7323a684 To: ;tag=as2e2179d6 Call-ID: 017fba790554a0ff20c5c7473d4edf6f@67.165.241.16 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 11 headers, 0 lines Destroying call '017fba790554a0ff20c5c7473d4edf6f@67.165.241.16' ast1000*CLI> sip no debug ast1000*CLI> SIP Debugging Disabled ast1000*CLI> exit Executing last minute cleanups