Asterisk SVN-trunk-r345023M, Copyright (C) 1999 - 2011 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details. ========================================================================= 'alwaysfork' is not compatible with console or remote console mode; ignored [Nov 14 14:26:54] Connected to Asterisk SVN-trunk-r345023M currently running on lx3 (pid = 17577) lx3*CLI> Verbosity is at least 1 lx3*CLI> core set verbose 9 lx3*CLI> Verbosity was 1 and is now 9 lx3*CLI> core set debug 9 lx3*CLI> Core debug was 0 and is now 9 lx3*CLI> sip set debug peer wimpys lx3*CLI> SIP Debugging Enabled for IP: 192.168.7.65 lx3*CLI> [2011-11-14 14:27:21] <--- SIP read from UDP:192.168.7.65:2072 ---> INVITE sip:11@sip.yeti.dk SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2072;branch=z9hG4bK-6qrds0q4whib;rport f: "WIMPy" ;tag=r6k2hy9k69 t: i: 3c27be4ea23e-oumydrpwtg4f CSeq: 1 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 c: application/sdp l: 810 v=0 o=root 1373301650 1373301650 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 62162 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0SrIh8+KWaoj8HhpA+6vYlBZ9PhjZoTlNVP08SbF a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 62162 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> lx3*CLI> [2011-11-14 14:27:21] --- (19 headers 34 lines) --- [2011-11-14 14:27:21] == Using UDPTL TOS bits 184 [2011-11-14 14:27:21] == Using UDPTL CoS mark 5 [2011-11-14 14:27:21] Sending to 192.168.7.65:2072 (no NAT) [2011-11-14 14:27:21] Using INVITE request as basis request - 3c27be4ea23e-oumydrpwtg4f [2011-11-14 14:27:21] Found peer 'wimpys' for 'wimpys' from 192.168.7.65:2072 [2011-11-14 14:27:21] <--- Reliably Transmitting (no NAT) to 192.168.7.65:2072 ---> SIP/2.0 401 Unauthorized v: SIP/2.0/UDP 192.168.7.65:2072;branch=z9hG4bK-6qrds0q4whib;received=192.168.7.65;rport=2072 f: "WIMPy" ;tag=r6k2hy9k69 t: ;tag=as46f82635 i: 3c27be4ea23e-oumydrpwtg4f CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="yeti.dk", nonce="192ee833" l: 0 <------------> lx3*CLI> [2011-11-14 14:27:21] Scheduling destruction of SIP dialog '3c27be4ea23e-oumydrpwtg4f' in 6400 ms (Method: INVITE) lx3*CLI> [2011-11-14 14:27:21] <--- SIP read from UDP:192.168.7.65:2072 ---> ACK sip:11@sip.yeti.dk SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2072;branch=z9hG4bK-6qrds0q4whib;rport f: "WIMPy" ;tag=r6k2hy9k69 t: ;tag=as46f82635 i: 3c27be4ea23e-oumydrpwtg4f CSeq: 1 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> lx3*CLI> [2011-11-14 14:27:21] --- (9 headers 0 lines) --- lx3*CLI> [2011-11-14 14:27:21] <--- SIP read from UDP:192.168.7.65:2072 ---> INVITE sip:11@sip.yeti.dk SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2072;branch=z9hG4bK-swqbgfte3bih;rport f: "WIMPy" ;tag=r6k2hy9k69 t: i: 3c27be4ea23e-oumydrpwtg4f CSeq: 2 INVITE Max-Forwards: 16 m: ;reg-id=1 X-Serialnumber: 000413234B25 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/8.4.27 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="wimpys",realm="yeti.dk",nonce="192ee833",uri="sip:11@sip.yeti.dk",response="90f39605ad2562a73458af17b2f6e5d0",algorithm=MD5 c: application/sdp l: 810 v=0 o=root 1373301650 1373301650 IN IP4 192.168.7.65 s=call c=IN IP4 192.168.7.65 t=0 0 m=audio 62162 RTP/SAVP 9 8 0 3 18 99 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0SrIh8+KWaoj8HhpA+6vYlBZ9PhjZoTlNVP08SbF a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv m=audio 62162 RTP/AVP 9 8 0 3 18 99 4 101 a=direction:both a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:10 a=sendrecv <-------------> [2011-11-14 14:27:21] --- (20 headers 34 lines) --- lx3*CLI> [2011-11-14 14:27:21] Sending to 192.168.7.65:2072 (no NAT) lx3*CLI> [2011-11-14 14:27:21] Using INVITE request as basis request - 3c27be4ea23e-oumydrpwtg4f lx3*CLI> [2011-11-14 14:27:21] Found peer 'wimpys' for 'wimpys' from 192.168.7.65:2072 lx3*CLI> [2011-11-14 14:27:21] == Using SIP RTP TOS bits 184 [2011-11-14 14:27:21] == Using SIP RTP CoS mark 5 lx3*CLI> [2011-11-14 14:27:21] Found RTP audio format 9 lx3*CLI> [2011-11-14 14:27:21] Found RTP audio format 8 lx3*CLI> [2011-11-14 14:27:21] Found RTP audio format 0 lx3*CLI> [2011-11-14 14:27:21] Found RTP audio format 3 lx3*CLI> [2011-11-14 14:27:21] Found RTP audio format 18 [2011-11-14 14:27:21] Found RTP audio format 99 lx3*CLI> [2011-11-14 14:27:21] Found RTP audio format 4 [2011-11-14 14:27:21] Found RTP audio format 101 lx3*CLI> [2011-11-14 14:27:21] Found audio description format G722 for ID 9 lx3*CLI> [2011-11-14 14:27:21] Found audio description format PCMA for ID 8 lx3*CLI> [2011-11-14 14:27:21] Found audio description format PCMU for ID 0 lx3*CLI> [2011-11-14 14:27:21] Found audio description format GSM for ID 3 lx3*CLI> [2011-11-14 14:27:21] Found audio description format G729 for ID 18 [2011-11-14 14:27:21] Found audio description format G726-32 for ID 99 lx3*CLI> [2011-11-14 14:27:21] Found audio description format G723 for ID 4 lx3*CLI> [2011-11-14 14:27:21] Found audio description format telephone-event for ID 101 lx3*CLI> [2011-11-14 14:27:21] WARNING[17791]: chan_sip.c:9023 process_sdp: Multiple audio streams are not supported lx3*CLI> [2011-11-14 14:27:21] <--- Reliably Transmitting (no NAT) to 192.168.7.65:2072 ---> SIP/2.0 488 Not acceptable here v: SIP/2.0/UDP 192.168.7.65:2072;branch=z9hG4bK-swqbgfte3bih;received=192.168.7.65;rport=2072 f: "WIMPy" ;tag=r6k2hy9k69 t: ;tag=as46f82635 i: 3c27be4ea23e-oumydrpwtg4f CSeq: 2 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH k: replaces, timer l: 0 <------------> [2011-11-14 14:27:21] Scheduling destruction of SIP dialog '3c27be4ea23e-oumydrpwtg4f' in 6400 ms (Method: INVITE) lx3*CLI> [2011-11-14 14:27:21] <--- SIP read from UDP:192.168.7.65:2072 ---> ACK sip:11@sip.yeti.dk SIP/2.0 v: SIP/2.0/UDP 192.168.7.65:2072;branch=z9hG4bK-swqbgfte3bih;rport f: "WIMPy" ;tag=r6k2hy9k69 t: ;tag=as46f82635 i: 3c27be4ea23e-oumydrpwtg4f CSeq: 2 ACK Max-Forwards: 16 m: ;reg-id=1 l: 0 <-------------> lx3*CLI> [2011-11-14 14:27:21] --- (9 headers 0 lines) --- lx3*CLI> [2011-11-14 14:27:23] Really destroying SIP dialog '3c271c44e97e-8o1o6fjiamfv' Method: SUBSCRIBE lx3*CLI> [2011-11-14 14:27:24] Really destroying SIP dialog '3c26701bbaeb-9wkqvkksuc2c' Method: REGISTER lx3*CLI> [2011-11-14 14:27:24] Really destroying SIP dialog '3c26701bb7ed-893o60xshpds' Method: REGISTER lx3*CLI> sip set deb[2011-11-14 14:27:28] Really destroying SIP dialog '3c27be4ea23e-oumydrpwtg4f' Method: ACK lx3*CLI> sip set debug off