Asterisk SVN-trunk-r46319M, 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. ========================================================================= Connected to Asterisk SVN-trunk-r46319M currently running on main (pid = 31390) main*CLI> Verbosity is at least 3 main*CLI> sip debug main*CLI> SIP Debugging enabled main*CLI> <--- SIP read from 74.52.15.138:5060 ---> INVITE sip:4156928449@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 74.52.15.138:5060;branch=z9hG4bK130d38fb;rport From: "Unavailable" ;tag=as7f649af5 To: Contact: Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 INVITE User-Agent: iCall Softswitch Max-Forwards: 70 Date: Mon, 11 Dec 2006 05:56:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 308 v=0 o=root 9124 9124 IN IP4 74.52.15.138 s=session c=IN IP4 74.52.15.138 t=0 0 m=audio 16712 RTP/AVP 0 18 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (13 headers 14 lines) --- Sending to 74.52.15.138 : 5060 (NAT) Using INVITE request as basis request - 10c504753292178630e25c257a8cd248@74.52.15.138 No user 'Restricted' in SIP users list Found peer 'icall' for 'Restricted' from 74.52.15.138:5060 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 74.52.15.138:16712 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format PCMA for ID 8 Found description format GSM for ID 3 Found description format telephone-event for ID 101 Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 74.52.15.138:16712 Looking for 4156928449 in from-icall (domain 198.144.201.82) list_route: hop: <--- Transmitting (no NAT) to 74.52.15.138:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 74.52.15.138:5060;branch=z9hG4bK130d38fb;rport;received=74.52.15.138 From: "Unavailable" ;tag=as7f649af5 To: Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 INVITE User-Agent: Caller Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> main*CLI> -- Executing [4156928449@from-icall:1] Goto("SIP/74.52.15.138-0972c8c8", "locals|298|1") in new stack -- Goto (locals,298,1) -- Executing [298@locals:1] Gosub("SIP/74.52.15.138-0972c8c8", "internal-ext|s|1(SIP/at320&SIP/brad)") in new stack -- Executing [s@internal-ext:1] Set("SIP/74.52.15.138-0972c8c8", "devext=SIP/at320&SIP/brad") in new stack -- Executing [s@internal-ext:2] Dial("SIP/74.52.15.138-0972c8c8", "SIP/at320&SIP/brad|90|t") in new stack [Dec 10 21:50:02] WARNING[8437]: translate.c:86 powerof: No bits set? 0 Audio is at 198.144.201.82 port 10350 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 68.122.32.101:5160: INVITE sip:at320@68.122.32.101:5160 SIP/2.0 Via: SIP/2.0/UDP 198.144.201.82:5160;branch=z9hG4bK1754b32b;rport From: "Unavailable" ;tag=as0f6ffc79 To: Contact: Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 102 INVITE User-Agent: Caller Asterisk Max-Forwards: 70 Date: Mon, 11 Dec 2006 05:50:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 301 v=0 o=root 31390 31390 IN IP4 198.144.201.82 s=session c=IN IP4 198.144.201.82 t=0 0 m=audio 10350 RTP/AVP 0 97 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called at320 [Dec 10 21:50:02] WARNING[8437]: translate.c:86 powerof: No bits set? 0 Audio is at 192.168.123.10 port 10788 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.123.18:5060: INVITE sip:192.168.123.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: Contact: Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 INVITE User-Agent: Caller Asterisk Max-Forwards: 70 Date: Mon, 11 Dec 2006 05:50:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 301 v=0 o=root 31390 31390 IN IP4 192.168.123.10 s=session c=IN IP4 192.168.123.10 t=0 0 m=audio 10788 RTP/AVP 0 97 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called brad main*CLI> <--- SIP read from 192.168.123.18:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- main*CLI> <--- SIP read from 192.168.123.18:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: ;tag=2ce6453bf4be04dc Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- main*CLI> -- SIP/brad-0976a7a0 is ringing main*CLI> <--- Transmitting (no NAT) to 74.52.15.138:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 74.52.15.138:5060;branch=z9hG4bK130d38fb;rport;received=74.52.15.138 From: "Unavailable" ;tag=as7f649af5 To: ;tag=as39e1a586 Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 INVITE User-Agent: Caller Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> main*CLI> <--- SIP read from 68.122.32.101:5160 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.144.201.82:5160;branch=z9hG4bK1754b32b;rport Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 102 INVITE From: "Unavailable" ;tag=as0f6ffc79 To: ;tag=ZmvkY0J3If2GtKse Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- main*CLI> -- SIP/at320-09749e28 is ringing main*CLI> <--- SIP read from 68.122.32.101:5160 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 198.144.201.82:5160;branch=z9hG4bK1754b32b;rport Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 102 INVITE From: "Unavailable" ;tag=as0f6ffc79 To: ;tag=ZmvkY0J3If2GtKse Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 194 v=0 o=- 19279402 60533422 IN IP4 68.122.32.101 s=SIP CALL c=IN IP4 68.122.32.101 t=0 0 m=audio 6001 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 68.122.32.101:6001 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x140c (ulaw|alaw|ilbc|g722), peer - audio=0x4 (ulaw)/video=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 68.122.32.101:6001 --- set_address_from_contact host '68.122.32.101' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 68.122.32.101, port 5160 Transmitting (no NAT) to 68.122.32.101:5160: ACK sip:at320@68.122.32.101:5160 SIP/2.0 Via: SIP/2.0/UDP 198.144.201.82:5160;branch=z9hG4bK5e60e202;rport From: "Unavailable" ;tag=as0f6ffc79 To: ;tag=ZmvkY0J3If2GtKse Contact: Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 102 ACK User-Agent: Caller Asterisk Max-Forwards: 70 Content-Length: 0 --- main*CLI> -- SIP/at320-09749e28 answered SIP/74.52.15.138-0972c8c8 Scheduling destruction of SIP dialog '071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.123.18:5060: CANCEL sip:192.168.123.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 CANCEL main*CLI> User-Agent: Caller Asterisk Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10' in 32000 ms (Method: INVITE) [Dec 10 21:50:04] WARNING[8437]: translate.c:86 powerof: No bits set? 0 [Dec 10 21:50:04] WARNING[8437]: translate.c:86 powerof: No bits set? 0 Audio is at 198.144.201.82 port 10546 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 74.52.15.138:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 74.52.15.138:5060;branch=z9hG4bK130d38fb;rport;received=74.52.15.138 From: "Unavailable" ;tag=as7f649af5 To: ;tag=as39e1a586 Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 INVITE User-Agent: Caller Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 323 v=0 o=root 31390 31390 IN IP4 68.122.32.101 s=session c=IN IP4 68.122.32.101 t=0 0 m=audio 6001 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv <------------> main*CLI> <--- SIP read from 192.168.123.18:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: ;tag=2ce6453bf4be04dc Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 CANCEL User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- main*CLI> <--- SIP read from 192.168.123.18:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: ;tag=2ce6453bf4be04dc Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.123.18:5060: ACK sip:192.168.123.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5160;branch=z9hG4bK2195a17d;rport From: "Unavailable" ;tag=as6ec47d44 To: ;tag=2ce6453bf4be04dc Contact: Call-ID: 071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10 CSeq: 102 ACK User-Agent: Caller Asterisk Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '071fcfe2092a40e121ae7b42205ea4c4@192.168.123.10' Method: INVITE main*CLI> <--- SIP read from 74.52.15.138:5060 ---> ACK sip:4156928449@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 74.52.15.138:5060;branch=z9hG4bK5e4f3d86;rport From: "Unavailable" ;tag=as7f649af5 To: ;tag=as39e1a586 Contact: Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 ACK User-Agent: iCall Softswitch Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> <--- SIP read from 68.122.32.101:5160 ---> INVITE sip:Restricted@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKGyPwVi0DcYX9gVzA;rport Max-Forwards: 70 User-Agent: IP Phone V1.54.004 CFG0 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 Contact: CSeq: 1 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 200 v=0 o=- 40730928 03331636 IN IP4 68.122.32.101 s=SIP CALL c=IN IP4 0.0.0.0 t=0 0 m=audio 6001 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> --- (12 headers 10 lines) --- Sending to 68.122.32.101 : 5160 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:6001 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x140c (ulaw|alaw|ilbc|g722), peer - audio=0x4 (ulaw)/video=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 0.0.0.0:6001 [Dec 10 21:50:08] WARNING[31464]: translate.c:86 powerof: No bits set? 0 Audio is at 198.144.201.82 port 10350 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 68.122.32.101:5160 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKGyPwVi0DcYX9gVzA;received=68.122.32.101;rport=5160 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 1 INVITE User-Agent: Caller Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 232 v=0 o=root 31390 31391 IN IP4 198.144.201.82 s=session c=IN IP4 198.144.201.82 t=0 0 m=audio 10350 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=recvonly <------------> main*CLI> -- Started music on hold, class 'default', on channel 'SIP/74.52.15.138-0972c8c8' main*CLI> <--- SIP read from 68.122.32.101:5160 ---> ACK sip:Restricted@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKyBh1L7hj9XSLCnjK;rport Max-Forwards: 70 User-Agent: IP Phone V1.54.004 CFG0 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 Contact: CSeq: 1 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- main*CLI> <--- SIP read from 68.122.32.101:5160 ---> INVITE sip:Restricted@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKgknyyAlhjOmW8MMr;rport Max-Forwards: 70 User-Agent: IP Phone V1.54.004 CFG0 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 Contact: CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 194 v=0 o=- 90871087 10668658 IN IP4 68.122.32.101 s=SIP CALL c=IN IP4 68.122.32.101 t=0 0 m=audio 6001 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (12 headers 9 lines) --- Sending to 68.122.32.101 : 5160 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 68.122.32.101:6001 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x140c (ulaw|alaw|ilbc|g722), peer - audio=0x4 (ulaw)/video=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 68.122.32.101:6001 [Dec 10 21:50:08] WARNING[31464]: translate.c:86 powerof: No bits set? 0 Audio is at 198.144.201.82 port 10350 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 68.122.32.101:5160 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKgknyyAlhjOmW8MMr;received=68.122.32.101;rport=5160 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 2 INVITE User-Agent: Caller Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 232 v=0 o=root 31390 31392 IN IP4 198.144.201.82 s=session c=IN IP4 198.144.201.82 t=0 0 m=audio 10350 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv <------------> main*CLI> -- Stopped music on hold on SIP/74.52.15.138-0972c8c8 main*CLI> <--- SIP read from 68.122.32.101:5160 ---> ACK sip:Restricted@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKV6wwMyBLZDaKcaQZ;rport Max-Forwards: 70 User-Agent: IP Phone V1.54.004 CFG0 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 Contact: CSeq: 2 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- main*CLI> <--- SIP read from 68.122.32.101:5160 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> main*CLI> <--- SIP read from 68.122.32.101:5160 ---> BYE sip:Restricted@198.144.201.82:5160 SIP/2.0 Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKwriAavBU6qVRrbwf;rport Max-Forwards: 70 User-Agent: IP Phone V1.54.004 CFG0 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 Contact: CSeq: 3 BYE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 68.122.32.101 : 5160 (NAT) <--- Transmitting (NAT) to 68.122.32.101:5160 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 68.122.32.101:5160;branch=z9hG4bKwriAavBU6qVRrbwf;received=68.122.32.101;rport=5160 From: ;tag=ZmvkY0J3If2GtKse To: "Unavailable" ;tag=as0f6ffc79 Call-ID: 49536a414251e4203a0097b7562b555f@198.144.201.82 CSeq: 3 BYE User-Agent: Caller Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (internal-ext, s, 2) exited non-zero on 'SIP/74.52.15.138-0972c8c8' Scheduling destruction of SIP dialog '10c504753292178630e25c257a8cd248@74.52.15.138' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 74.52.15.138, port 5060 Reliably Transmitting (no NAT) to 74.52.15.138:5060: BYE sip:Restricted@74.52.15.138 SIP/2.0 Via: SIP/2.0/UDP 198.144.201.82:5160;branch=z9hG4bK5548e6a0;rport From: ;tag=as39e1a586 To: "Unavailable" ;tag=as7f649af5 Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 BYE User-Agent: Caller Asterisk Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '49536a414251e4203a0097b7562b555f@198.144.201.82' Method: BYE main*CLI> <--- SIP read from 74.52.15.138:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 198.144.201.82:5160;branch=z9hG4bK5548e6a0;received=198.144.201.82;rport=5160 From: ;tag=as39e1a586 To: "Unavailable" ;tag=as7f649af5 Call-ID: 10c504753292178630e25c257a8cd248@74.52.15.138 CSeq: 102 BYE User-Agent: iCall Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '10c504753292178630e25c257a8cd248@74.52.15.138' Method: ACK main*CLI> exit