Asterisk 1.4.0-beta3, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.0-beta3 currently running on mythaster (pid = 4892) mythaster*CLI> Verbosity is at least 100 [Kmythaster*CLI> sip debug mythaster*CLI> SIP Debugging enabled [Kmythaster*CLI> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '21fe35a7456bb76d68d4cef432d071ad@127.0.1.1' in 32000 ms (Method: REGISTER) [Nov 20 13:24:30] [1;33;40mNOTICE[0;37;40m[4916]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m11771[0;37;40m [1;37;40mhandle_response_register[0;37;40m: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s) [Kmythaster*CLI> Really destroying SIP dialog '21fe35a7456bb76d68d4cef432d071ad@127.0.1.1' Method: REGISTER [Kmythaster*CLI> <--- SIP read from 192.168.1.3:5060 ---> INVITE sip:7003@192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-96abf460 From: 7002 ;tag=5b71af29ea245478o1 To: Call-ID: 28bf5bd-e8929ef4@192.168.1.3 CSeq: 101 INVITE Max-Forwards: 70 Contact: 7002 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 110350 110350 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16464 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 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:30 a=sendrecv <-------------> --- (14 headers 19 lines) --- Sending to 192.168.1.3 : 5060 (no NAT) Using INVITE request as basis request - 28bf5bd-e8929ef4@192.168.1.3 Found peer '7003' <--- Reliably Transmitting (no NAT) to 192.168.1.3:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-96abf460;received=192.168.1.3 From: 7002 ;tag=5b71af29ea245478o1 To: ;tag=as0ccfc6c8 Call-ID: 28bf5bd-e8929ef4@192.168.1.3 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f7dd2f9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '28bf5bd-e8929ef4@192.168.1.3' in 32000 ms (Method: INVITE) [Kmythaster*CLI> <--- SIP read from 192.168.1.3:5060 ---> ACK sip:7003@192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-96abf460 From: 7002 ;tag=5b71af29ea245478o1 To: ;tag=as0ccfc6c8 Call-ID: 28bf5bd-e8929ef4@192.168.1.3 CSeq: 101 ACK Max-Forwards: 70 Contact: 7002 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kmythaster*CLI> <--- SIP read from 192.168.1.3:5060 ---> INVITE sip:7003@192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-6ad9a8ac From: 7002 ;tag=5b71af29ea245478o1 To: Call-ID: 28bf5bd-e8929ef4@192.168.1.3 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="7002",realm="asterisk",nonce="5f7dd2f9",uri="sip:7003@192.168.1.4",algorithm=MD5,response="47c902d44efa3dbad4d1a63a9cb4074f" Contact: 7002 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 110350 110350 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16464 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 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:30 a=sendrecv <-------------> --- (15 headers 19 lines) --- Sending to 192.168.1.3 : 5060 (no NAT) Using INVITE request as basis request - 28bf5bd-e8929ef4@192.168.1.3 Found peer '7003' [Nov 20 13:24:39] [1;31;40mWARNING[0;37;40m[4916]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m7916[0;37;40m [1;37;40mcheck_auth[0;37;40m: username mismatch, have <7003>, digest has <7002> [Nov 20 13:24:39] [1;33;40mNOTICE[0;37;40m[4916]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m13017[0;37;40m [1;37;40mhandle_request_invite[0;37;40m: Failed to authenticate user 7002 ;tag=5b71af29ea245478o1 <--- Reliably Transmitting (no NAT) to 192.168.1.3:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-6ad9a8ac;received=192.168.1.3 From: 7002 ;tag=5b71af29ea245478o1 To: ;tag=as0ccfc6c8 Call-ID: 28bf5bd-e8929ef4@192.168.1.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '28bf5bd-e8929ef4@192.168.1.3' in 32000 ms (Method: INVITE) [Kmythaster*CLI> <--- SIP read from 192.168.1.3:5060 ---> ACK sip:7003@192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-6ad9a8ac From: 7002 ;tag=5b71af29ea245478o1 To: ;tag=as0ccfc6c8 Call-ID: 28bf5bd-e8929ef4@192.168.1.3 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="7002",realm="asterisk",nonce="5f7dd2f9",uri="sip:7003@192.168.1.4",algorithm=MD5,response="5807a20a5d997d5bf5532460f0417e43" Contact: 7002 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 0 <-------------> [Kmythaster*CLI> --- (11 headers 0 lines) --- [Kmythaster*CLI> sip no debug mythaster*CLI> SIP Debugging Disabled [Kmythaster*CLI> exit Executing last minute cleanups