Asterisk 1.6.0.1, Copyright (C) 1999 - 2008 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found Connected to Asterisk 1.6.0.1 currently running on asterisk (pid = 4055) asterisk*CLI> Verbosity was 3 and is now 4 Core debug is at least 1 asterisk*CLI> Reliably Transmitting (NAT) to 193.111.200.56:5060: OPTIONS sip:sip.gradwell.net SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK4ec9e2a3;rport Max-Forwards: 70 From: "asterisk" ;tag=as52930589 To: Contact: Call-ID: 7c2b48ec6bad6ae91eeee89519674987@123.456.123.456 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:38:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK4ec9e2a3;rport=5060 From: "asterisk" ;tag=as52930589 To: ;tag=95e55e7393f7733fb0d79381494abb70.d3d6 Call-ID: 7c2b48ec6bad6ae91eeee89519674987@123.456.123.456 CSeq: 102 OPTIONS Server: Sip EXpress router (0.9.4 (i386/freebsd)) Content-Length: 0 Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=56371 req_src_ip=123.456.123.456 req_src_port=5060 in_uri=sip:sip.gradwell.net out_uri=sip:sip.gradwell.net via_cnt==1" <-------------> asterisk*CLI> --- (9 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '7c2b48ec6bad6ae91eeee89519674987@123.456.123.456' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> INVITE sip:90800500005@asterisk;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-0i8q0ttj7sfk;rport From: "Matt Brewster" ;tag=vvm7buitjp To: Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/7.1.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 304 v=0 o=root 1306287720 1306287720 IN IP4 172.25.0.200 s=call c=IN IP4 172.25.0.200 t=0 0 m=audio 58306 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+yYMw4BXph0vf8tvZsJphlNt77wlvCHTQPvKq+QT a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (18 headers 12 lines) --- == Using SIP RTP CoS mark 5 Sending to 172.25.0.200 : 1036 (NAT) Using INVITE request as basis request - 3c3108967f2c-df55m86jnd5v Found user '1001' for '1001' Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 172.25.0.200:58306 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.0.200:58306 Looking for 90800500005 in trusted_phones (domain asterisk) list_route: hop: <--- Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-0i8q0ttj7sfk;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> -- Executing [90800500005@trusted_phones:1] Verbose("SIP/1001-0823e630", "!! "Matt Brewster" <1001> making outgoing call to 0800500005 via Gradwell SIP trunk") in new stack asterisk*CLI> !! Matt Brewster <1001> making outgoing call to 0800500005 via Gradwell SIP trunk asterisk*CLI> -- Executing [90800500005@trusted_phones:2] Set("SIP/1001-0823e630", "CALLERID(num)=023023123456") in new stack asterisk*CLI> -- Executing [90800500005@trusted_phones:3] Dial("SIP/1001-0823e630", "SIP/gradwell_out/0800500005") in new stack asterisk*CLI> == Using SIP RTP CoS mark 5 asterisk*CLI> Audio is at 123.456.123.456 port 16434 asterisk*CLI> Adding codec 0x8 (alaw) to SDP asterisk*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> Reliably Transmitting (NAT) to 193.111.200.56:5060: INVITE sip:0800500005@sip.gradwell.net SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK25a41eaf;rport Max-Forwards: 70 From: "Matt Brewster" ;tag=as7ef7abb3 To: Contact: Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:38:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 715639994 715639994 IN IP4 123.456.123.456 s=Asterisk PBX 1.6.0.1 c=IN IP4 123.456.123.456 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI> -- Called gradwell_out/0800500005 asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK25a41eaf;rport=5060 From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=95e55e7393f7733fb0d79381494abb70.55bd Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sip.gradwell.net", nonce="49105171c3e9cc05727e5893f3bdff5ade410603", qop="auth" Server: Sip EXpress router (0.9.4 (i386/freebsd)) Content-Length: 0 Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=56371 req_src_ip=123.456.123.456 req_src_port=5060 in_uri=sip:0800500005@sip.gradwell.net out_uri=sip:0800500005@sip.gradwell.net via_cnt==1" <-------------> --- (10 headers 0 lines) --- asterisk*CLI> Transmitting (NAT) to 193.111.200.56:5060: ACK sip:0800500005@sip.gradwell.net SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK25a41eaf;rport Max-Forwards: 70 From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=95e55e7393f7733fb0d79381494abb70.55bd Contact: Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 --- Audio is at 123.456.123.456 port 16434 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 193.111.200.56:5060: INVITE sip:0800500005@sip.gradwell.net SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK5e251b56;rport Max-Forwards: 70 From: "Matt Brewster" ;tag=as7ef7abb3 To: Contact: Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.1 Proxy-Authorization: Digest username="123456", realm="sip.gradwell.net", algorithm=MD5, uri="sip:0800500005@sip.gradwell.net", nonce="49105171c3e9cc05727e5893f3bdff5ade410603", response="4f3f8af40d1a6d238a5bc941db2a3229", qop=auth, cnonce="07573c2c", nc=00000001 Date: Tue, 04 Nov 2008 13:38:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 715639994 715639995 IN IP4 123.456.123.456 s=Asterisk PBX 1.6.0.1 c=IN IP4 123.456.123.456 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK5e251b56;rport=5060 From: "Matt Brewster" ;tag=as7ef7abb3 To: Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 103 INVITE Server: Sip EXpress router (0.9.4 (i386/freebsd)) Content-Length: 0 Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=56370 req_src_ip=123.456.123.456 req_src_port=5060 in_uri=sip:0800500005@sip.gradwell.net out_uri=sip:*770800500005@sip-prepay.lb.gradwell.net:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK5e251b56;rport=5060 From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=as64ba6fa3 Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk*CLI> -- SIP/gradwell_out-08230d78 is ringing asterisk*CLI> <--- Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-0i8q0ttj7sfk;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK5e251b56;rport=5060 From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=as64ba6fa3 Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 222 v=0 o=root 20381 20381 IN IP4 193.111.200.132 s=session c=IN IP4 193.111.200.132 t=0 0 m=audio 15292 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (11 headers 10 lines) --- asterisk*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 193.111.200.132:15292 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.111.200.132:15292 -- SIP/gradwell_out-08230d78 is making progress passing it to SIP/1001-0823e630 asterisk*CLI> Audio is at 172.25.0.1 port 15080 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-0i8q0ttj7sfk;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 2092525023 2092525023 IN IP4 172.25.0.1 s=Asterisk PBX 1.6.0.1 c=IN IP4 172.25.0.1 t=0 0 m=audio 15080 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK5e251b56;rport=5060 Record-Route: From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=as64ba6fa3 Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 222 v=0 o=root 20381 20382 IN IP4 193.111.200.132 s=session c=IN IP4 193.111.200.132 t=0 0 m=audio 15292 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (12 headers 10 lines) --- asterisk*CLI> Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 193.111.200.132:15292 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 193.111.200.132:15292 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 193.111.200.56, port 5060 Transmitting (NAT) to 193.111.200.56:5060: ACK sip:*770800500005@193.111.200.132 SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK47d1a568;rport Route: Max-Forwards: 70 From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=as64ba6fa3 Contact: Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 --- -- SIP/gradwell_out-08230d78 answered SIP/1001-0823e630 Audio is at 172.25.0.1 port 15080 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-0i8q0ttj7sfk;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 2092525023 2092525024 IN IP4 172.25.0.1 s=Asterisk PBX 1.6.0.1 c=IN IP4 172.25.0.1 t=0 0 m=audio 15080 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> asterisk*CLI> -- Packet2Packet bridging SIP/1001-0823e630 and SIP/gradwell_out-08230d78 asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> ACK sip:90800500005@172.25.0.1 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-wg388lswtnor;rport From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> asterisk*CLI> --- (9 headers 0 lines) --- asterisk*CLI> Reliably Transmitting (NAT) to 172.25.0.198:1036: OPTIONS sip:1000@172.25.0.198:1036;line=a8cyy86p SIP/2.0 Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK07c96c14;rport Max-Forwards: 70 From: "asterisk" ;tag=as0ce13cbc To: Contact: Call-ID: 6ee031871da12c4a3178481534e87ca4@172.25.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:38:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://172.25.0.198:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK07c96c14;rport=5060 From: "asterisk" ;tag=as0ce13cbc To: Call-ID: 6ee031871da12c4a3178481534e87ca4@172.25.0.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom370/7.1.35 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '6ee031871da12c4a3178481534e87ca4@172.25.0.1' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> INVITE sip:90800500005@172.25.0.1 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-e3do48q4aifb;rport From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/7.1.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 220 v=0 o=root 1306287720 1306287721 IN IP4 172.25.0.200 s=call c=IN IP4 172.25.0.200 t=0 0 m=audio 58306 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> --- (18 headers 11 lines) --- Sending to 172.25.0.200 : 1036 (NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 172.25.0.200:58306 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.0.200:58306 <--- Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-e3do48q4aifb;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 172.25.0.1 port 15080 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-e3do48q4aifb;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 2092525023 2092525025 IN IP4 172.25.0.1 s=Asterisk PBX 1.6.0.1 c=IN IP4 172.25.0.1 t=0 0 m=audio 15080 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> asterisk*CLI> -- Started music on hold, class 'default', on SIP/gradwell_out-08230d78 asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> ACK sip:90800500005@172.25.0.1 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-csu3kun0aa82;rport From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- RTCP SR transmission error, rtcp halted asterisk*CLI> Reliably Transmitting (NAT) to 172.25.0.199:1036: OPTIONS sip:1003@172.25.0.199:1036;line=2ur8wqgt SIP/2.0 Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK2b479528;rport Max-Forwards: 70 From: "asterisk" ;tag=as58e85442 To: Contact: Call-ID: 489f04374216506a08816fa544450ca1@172.25.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:38:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://172.25.0.199:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK2b479528;rport=5060 From: "asterisk" ;tag=as58e85442 To: Call-ID: 489f04374216506a08816fa544450ca1@172.25.0.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom370/7.1.35 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '489f04374216506a08816fa544450ca1@172.25.0.1' Method: OPTIONS asterisk*CLI> RTCP SR transmission error, rtcp halted asterisk*CLI> Reliably Transmitting (NAT) to 172.25.0.197:1028: OPTIONS sip:1002@172.25.0.197:1028;line=2c55p9r3 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK16d77ab2;rport Max-Forwards: 70 From: "asterisk" ;tag=as5e1282eb To: Contact: Call-ID: 50d997cc5d120ce138e16f6f7c49149d@172.25.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:38:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://172.25.0.197:1028 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK16d77ab2;rport=5060 From: "asterisk" ;tag=as5e1282eb To: Call-ID: 50d997cc5d120ce138e16f6f7c49149d@172.25.0.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom370/7.1.35 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '50d997cc5d120ce138e16f6f7c49149d@172.25.0.1' Method: OPTIONS asterisk*CLI> Reliably Transmitting (NAT) to 172.25.0.200:1036: OPTIONS sip:1001@172.25.0.200:1036;line=0n301nlg SIP/2.0 Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK54643131;rport Max-Forwards: 70 From: "asterisk" ;tag=as690ff8ca To: Contact: Call-ID: 5507e4ab7c57c591401225fb4f5a5f0b@172.25.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:38:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.1:5060;branch=z9hG4bK54643131;rport=5060 From: "asterisk" ;tag=as690ff8ca To: Call-ID: 5507e4ab7c57c591401225fb4f5a5f0b@172.25.0.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom370/7.1.35 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> asterisk*CLI> --- (14 headers 0 lines) --- Really destroying SIP dialog '5507e4ab7c57c591401225fb4f5a5f0b@172.25.0.1' Method: OPTIONS RTCP SR transmission error, rtcp halted asterisk*CLI> RTCP SR transmission error, rtcp halted asterisk*CLI> -- ast_get_srv: SRV lookup for '_sip._UDP.sip.gradwell.net' mapped to host sip.gradwell.net, port 5060 asterisk*CLI> REGISTER 13 headers, 0 lines asterisk*CLI> Reliably Transmitting (NAT) to 193.111.200.56:5060: REGISTER sip:sip.gradwell.net SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK3de8a63e;rport Max-Forwards: 70 From: ;tag=as4db95c6c To: Call-ID: 3ba7b9035a15e83817eb9f117e2d3530@10.100.100.206 CSeq: 7809 REGISTER User-Agent: Asterisk PBX 1.6.0.1 Authorization: Digest username="123456", realm="sip.gradwell.net", algorithm=MD5, uri="sip:sip.gradwell.net", nonce="491051309fc56497d681b1d02a9f1082dda9df0b", response="45c2a7a6bbb9480d4aff507229391ade", qop=auth, cnonce="12179694", nc=00000002 Expires: 120 Contact: Event: registration Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK3de8a63e;rport=5060 From: ;tag=as4db95c6c To: ;tag=95e55e7393f7733fb0d79381494abb70.12bb Call-ID: 3ba7b9035a15e83817eb9f117e2d3530@10.100.100.206 CSeq: 7809 REGISTER Contact: ;expires=120 Server: Sip EXpress router (0.9.4 (i386/freebsd)) Content-Length: 0 Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=56371 req_src_ip=123.456.123.456 req_src_port=5060 in_uri=sip:sip.gradwell.net out_uri=sip:sip.gradwell.net via_cnt==1" <-------------> --- (10 headers 0 lines) --- asterisk*CLI> Scheduling destruction of SIP dialog '3ba7b9035a15e83817eb9f117e2d3530@10.100.100.206' in 32000 ms (Method: REGISTER) asterisk*CLI> RTCP SR transmission error, rtcp halted asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> INVITE sip:90800500005@172.25.0.1 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-xupnygn8pi7x;rport From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 3 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/7.1.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 220 v=0 o=root 1306287720 1306287722 IN IP4 172.25.0.200 s=call c=IN IP4 172.25.0.200 t=0 0 m=audio 58306 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (18 headers 11 lines) --- Sending to 172.25.0.200 : 1036 (NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 172.25.0.200:58306 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.0.200:58306 <--- Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-xupnygn8pi7x;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 3 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 172.25.0.1 port 15080 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-xupnygn8pi7x;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 3 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 2092525023 2092525026 IN IP4 172.25.0.1 s=Asterisk PBX 1.6.0.1 c=IN IP4 172.25.0.1 t=0 0 m=audio 15080 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> asterisk*CLI> -- Stopped music on hold on SIP/gradwell_out-08230d78 asterisk*CLI> <--- SIP read from UDP://172.25.0.200:1036 ---> ACK sip:90800500005@172.25.0.1 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-bmbsvs9lqaj9;rport From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 3 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP://172.25.0.200:1036 ---> BYE sip:90800500005@172.25.0.1 SIP/2.0 Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-q85q6783lya4;rport From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 4 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom370/7.1.35 RTP-RxStat: Total_Rx_Pkts=658,Rx_Pkts=164,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=744,Tx_Pkts=158,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> asterisk*CLI> --- (12 headers 0 lines) --- asterisk*CLI> Sending to 172.25.0.200 : 1036 (NAT) <--- Transmitting (NAT) to 172.25.0.200:1036 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.0.200:1036;branch=z9hG4bK-q85q6783lya4;received=172.25.0.200;rport=1036 From: "Matt Brewster" ;tag=vvm7buitjp To: ;tag=as191f8691 Call-ID: 3c3108967f2c-df55m86jnd5v CSeq: 4 BYE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> Scheduling destruction of SIP dialog '02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net' in 6400 ms (Method: INVITE) asterisk*CLI> set_destination: Parsing for address/port to send to asterisk*CLI> set_destination: set destination to 193.111.200.56, port 5060 asterisk*CLI> Reliably Transmitting (NAT) to 193.111.200.56:5060: BYE sip:*770800500005@193.111.200.132 SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK23a5d2b5;rport Route: Max-Forwards: 70 From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=as64ba6fa3 Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 104 BYE User-Agent: Asterisk PBX 1.6.0.1 Proxy-Authorization: Digest username="123456", realm="sip.gradwell.net", algorithm=MD5, uri="sip:*770800500005@193.111.200.132", nonce="49105171c3e9cc05727e5893f3bdff5ade410603", response="ca94d39411626229eadf52c3235fadcd", qop=auth, cnonce="0cfb2ce6", nc=00000002 Content-Length: 0 --- asterisk*CLI> == Spawn extension (trusted_phones, 90800500005, 3) exited non-zero on 'SIP/1001-0823e630' asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK23a5d2b5;rport=5060 Record-Route: From: "Matt Brewster" ;tag=as7ef7abb3 To: ;tag=as64ba6fa3 Call-ID: 02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing <-------------> asterisk*CLI> --- (12 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '02f6aebb3d7338fa09b56bee1b83a683@sip.gradwell.net' Method: INVITE asterisk*CLI> Really destroying SIP dialog '3c3108967f2c-df55m86jnd5v' Method: BYE asterisk*CLI> Reliably Transmitting (NAT) to 87.194.132.4:5060: OPTIONS sip:2001@172.30.0.150:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK3e4a8c28;rport Max-Forwards: 70 From: "asterisk" ;tag=as3d8ad30f To: Contact: Call-ID: 1d81476a7918e24450a9c426679398c9@123.456.123.456 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:39:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://87.194.132.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK3e4a8c28;rport From: "asterisk" ;tag=as3d8ad30f To: ;tag=8d1f2a0b8f935b72 Call-ID: 1d81476a7918e24450a9c426679398c9@123.456.123.456 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.6.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> asterisk*CLI> --- (11 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '1d81476a7918e24450a9c426679398c9@123.456.123.456' Method: OPTIONS asterisk*CLI> Reliably Transmitting (NAT) to 193.111.200.56:5060: OPTIONS sip:sip.gradwell.net SIP/2.0 Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK0bf00a27;rport Max-Forwards: 70 From: "asterisk" ;tag=as31544f74 To: Contact: Call-ID: 1e122b30721d88b253a90d077c6a5b25@123.456.123.456 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.1 Date: Tue, 04 Nov 2008 13:39:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://193.111.200.56:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 123.456.123.456:5060;branch=z9hG4bK0bf00a27;rport=5060 From: "asterisk" ;tag=as31544f74 To: ;tag=95e55e7393f7733fb0d79381494abb70.5619 Call-ID: 1e122b30721d88b253a90d077c6a5b25@123.456.123.456 CSeq: 102 OPTIONS Server: Sip EXpress router (0.9.4 (i386/freebsd)) Content-Length: 0 Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=56370 req_src_ip=123.456.123.456 req_src_port=5060 in_uri=sip:sip.gradwell.net out_uri=sip:sip.gradwell.net via_cnt==1" <-------------> asterisk*CLI> --- (9 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '1e122b30721d88b253a90d077c6a5b25@123.456.123.456' Method: OPTIONS asterisk*CLI> exit Executing last minute cleanups