[May 18 08:47:51] Asterisk 13.13.1 built by root @ asterisk.p98.belkam.com on a x86_64 running Linux on 2016-12-21 11:19:01 UTC [May 18 08:47:51] VERBOSE[5643] config.c: Parsing '/etc/asterisk/logger.conf': Found [May 18 08:47:52] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-de8cf1f2 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: dc806fc6-1ea479eb@192.168.105.20 CSeq: 1001 SUBSCRIBE Max-Forwards: 70 Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:47:52] VERBOSE[22763] chan_sip.c: --- (12 headers 0 lines) --- [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:47:52] VERBOSE[22763] sip/route.c: sip_route_dump: route/path hop: [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:47:52] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-de8cf1f2;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as4d30882e Call-ID: dc806fc6-1ea479eb@192.168.105.20 CSeq: 1001 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c17677d" Content-Length: 0 <------------> [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog 'dc806fc6-1ea479eb@192.168.105.20' in 6400 ms (Method: SUBSCRIBE) [May 18 08:47:52] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-27178e06 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: dc806fc6-1ea479eb@192.168.105.20 CSeq: 1002 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="6051",realm="asterisk",nonce="5c17677d",uri="sip:asterisk.p98.belkam.com",algorithm=MD5,response="4ea0adb010673e7187b6260b737bd199" Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:47:52] VERBOSE[22763] chan_sip.c: --- (13 headers 0 lines) --- [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:47:52] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-27178e06;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as4d30882e Call-ID: dc806fc6-1ea479eb@192.168.105.20 CSeq: 1002 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:47:52] VERBOSE[22763] chan_sip.c: Really destroying SIP dialog 'dc806fc6-1ea479eb@192.168.105.20' Method: SUBSCRIBE [May 18 08:47:53] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.26.116:5060 ---> NOTIFY sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.26.116:5060;branch=z9hG4bK-e27419b1 From: "6054" ;tag=69972247ad9fa1c3o0 To: Call-ID: 79bb9717-f8113333@192.168.26.116 CSeq: 5916 NOTIFY Max-Forwards: 70 Contact: "6054" Event: keep-alive User-Agent: Linksys/SPA922-6.1.5(a) Content-Length: 0 <-------------> [May 18 08:47:53] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:47:53] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.26.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.116:5060;branch=z9hG4bK-e27419b1;received=192.168.26.116 From: "6054" ;tag=69972247ad9fa1c3o0 To: ;tag=as43ab47f2 Call-ID: 79bb9717-f8113333@192.168.26.116 CSeq: 5916 NOTIFY Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:47:53] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog '79bb9717-f8113333@192.168.26.116' in 32000 ms (Method: NOTIFY) [May 18 08:47:53] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:47:53] VERBOSE[6436] asterisk.c: Remote UNIX connection disconnected [May 18 08:47:55] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:47:55] VERBOSE[6470] asterisk.c: Remote UNIX connection disconnected [May 18 08:47:56] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> INVITE sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 INVITE Max-Forwards: 70 Contact: "6053" Expires: 240 User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 301205 301205 IN IP4 192.168.158.20 s=- c=IN IP4 192.168.158.20 t=0 0 m=audio 16418 RTP/AVP 18 0 2 8 9 96 97 98 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [May 18 08:47:56] VERBOSE[22763] chan_sip.c: --- (14 headers 18 lines) --- [May 18 08:47:56] VERBOSE[22763] chan_sip.c: Sending to 192.168.158.20:5060 (no NAT) [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Sending to 192.168.158.20:5060 (no NAT) [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Using INVITE request as basis request - 4bd677e5-d5ccac73@192.168.158.20 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found peer '6053' for '6053' from 192.168.158.20:5060 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.158.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as68cba49b Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="665e2444" Content-Length: 0 <------------> [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Scheduling destruction of SIP dialog '4bd677e5-d5ccac73@192.168.158.20' in 6400 ms (Method: INVITE) [May 18 08:47:56] VERBOSE[22763] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.158.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as68cba49b Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="665e2444" Content-Length: 0 --- [May 18 08:47:56] VERBOSE[22763] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.158.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as68cba49b Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="665e2444" Content-Length: 0 --- [May 18 08:47:56] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> ACK sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as68cba49b Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 ACK Max-Forwards: 70 Contact: "6053" User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 0 <-------------> [May 18 08:47:56] VERBOSE[22763] chan_sip.c: --- (10 headers 0 lines) --- [May 18 08:47:56] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> INVITE sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="6053",realm="asterisk",nonce="665e2444",uri="sip:3550@asterisk.p98.belkam.com",algorithm=MD5,response="0fcedcc7874c35e5c304d24a13d745ed" Contact: "6053" Expires: 240 User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 301205 301205 IN IP4 192.168.158.20 s=- c=IN IP4 192.168.158.20 t=0 0 m=audio 16418 RTP/AVP 18 0 2 8 9 96 97 98 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [May 18 08:47:56] VERBOSE[22763] chan_sip.c: --- (15 headers 18 lines) --- [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Sending to 192.168.158.20:5060 (no NAT) [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Using INVITE request as basis request - 4bd677e5-d5ccac73@192.168.158.20 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found peer '6053' for '6053' from 192.168.158.20:5060 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] netsock2.c: Using SIP RTP TOS bits 184 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] netsock2.c: Using SIP RTP CoS mark 5 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 18 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 0 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 2 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 8 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 9 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 96 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 97 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 98 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found RTP audio format 101 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found audio description format G729a for ID 18 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found audio description format PCMU for ID 0 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found audio description format G726-32 for ID 2 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found audio description format PCMA for ID 8 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found audio description format G722 for ID 9 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found unknown media description format G726-40 for ID 96 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found unknown media description format G726-24 for ID 97 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found unknown media description format G726-16 for ID 98 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Found audio description format telephone-event for ID 101 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Capabilities: us - (g729), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (g729) [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Peer audio RTP is at port 192.168.158.20:16418 [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: Looking for 3550 in sipphones (domain asterisk.p98.belkam.com) [May 18 08:47:56] VERBOSE[22763][C-0000bd27] sip/route.c: sip_route_dump: route/path hop: [May 18 08:47:56] VERBOSE[22763][C-0000bd27] chan_sip.c: <--- Transmitting (no NAT) to 192.168.158.20:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [May 18 08:47:56] VERBOSE[6498][C-0000bd27] pbx.c: Executing [3550@sipphones:1] Dial("SIP/6053-00002a0d", "OOH323/3550") in new stack [May 18 08:47:56] VERBOSE[6498][C-0000bd27] app_dial.c: Called OOH323/3550 [May 18 08:47:56] VERBOSE[22763] chan_sip.c: Really destroying SIP dialog '7697458b100df50b0e663ef041489b99@192.168.72.254:5060' Method: BYE [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> ACK sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as68cba49b Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 ACK Max-Forwards: 70 Contact: "6053" User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 0 <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (10 headers 0 lines) --- [May 18 08:47:57] VERBOSE[6498][C-0000bd27] app_dial.c: Everyone is busy/congested at this time (1:1/0/0) [May 18 08:47:57] VERBOSE[6498][C-0000bd27] pbx.c: Executing [3550@sipphones:2] Hangup("SIP/6053-00002a0d", "") in new stack [May 18 08:47:57] VERBOSE[6498][C-0000bd27] pbx.c: Spawn extension (sipphones, 3550, 2) exited non-zero on 'SIP/6053-00002a0d' [May 18 08:47:57] VERBOSE[6498][C-0000bd27] chan_sip.c: Scheduling destruction of SIP dialog '4bd677e5-d5ccac73@192.168.158.20' in 6400 ms (Method: INVITE) [May 18 08:47:57] VERBOSE[6498][C-0000bd27] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.158.20:5060 ---> SIP/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as0bb7c79c Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> ACK sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-733c65d3 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as68cba49b Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 101 ACK Max-Forwards: 70 Contact: "6053" User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 0 <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (10 headers 0 lines) --- [May 18 08:47:57] VERBOSE[22763] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.158.20:5060: SIP/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as0bb7c79c Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.72.254:5060 ---> INVITE sip:6000@192.168.22.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK11bac3a3 Max-Forwards: 70 From: "D.Melekhov" ;tag=as41717c2d To: Contact: Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 18 May 2017 04:47:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "D.Melekhov" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 268 v=0 o=root 1160209453 1160209453 IN IP4 192.168.72.254 s=Asterisk PBX 13.13.1 c=IN IP4 192.168.72.254 t=0 0 m=audio 30314 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (15 headers 12 lines) --- [May 18 08:47:57] VERBOSE[22763] chan_sip.c: Sending to 192.168.72.254:5060 (no NAT) [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Sending to 192.168.72.254:5060 (no NAT) [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Using INVITE request as basis request - 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found peer 'ast-lud' for '6401' from 192.168.72.254:5060 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] netsock2.c: Using SIP RTP TOS bits 184 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] netsock2.c: Using SIP RTP CoS mark 5 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found RTP audio format 8 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found RTP audio format 0 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found RTP audio format 101 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found audio description format PCMA for ID 8 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found audio description format PCMU for ID 0 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Found audio description format telephone-event for ID 101 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw) [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Peer audio RTP is at port 192.168.72.254:30314 [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: Looking for 6000 in h323 (domain 192.168.22.19) [May 18 08:47:57] VERBOSE[22763][C-0000bd28] sip/route.c: sip_route_dump: route/path hop: [May 18 08:47:57] VERBOSE[22763][C-0000bd28] chan_sip.c: <--- Transmitting (no NAT) to 192.168.72.254:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK11bac3a3;received=192.168.72.254 From: "D.Melekhov" ;tag=as41717c2d To: Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@h323:1] Goto("SIP/ast-lud-00002a0e", "meetmectx,6000,1") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx_builtins.c: Goto (meetmectx,6000,1) [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:1] Set("SIP/ast-lud-00002a0e", "CHANNEL(language)=ru") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:2] Set("SIP/ast-lud-00002a0e", "CONNECTEDLINE(name,i)=Conf. 6000") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:3] Set("SIP/ast-lud-00002a0e", "CONNECTEDLINE(pres)=allowed") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:4] GotoIf("SIP/ast-lud-00002a0e", " 0?:havename") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx_builtins.c: Goto (meetmectx,6000,6) [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:6] Ringing("SIP/ast-lud-00002a0e", "") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] chan_sip.c: <--- Transmitting (no NAT) to 192.168.72.254:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK11bac3a3;received=192.168.72.254 From: "D.Melekhov" ;tag=as41717c2d To: ;tag=as5380c836 Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:7] Answer("SIP/ast-lud-00002a0e", "") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] chan_sip.c: Audio is at 30412 [May 18 08:47:57] VERBOSE[6508][C-0000bd28] chan_sip.c: Adding codec alaw to SDP [May 18 08:47:57] VERBOSE[6508][C-0000bd28] chan_sip.c: Adding codec ulaw to SDP [May 18 08:47:57] VERBOSE[6508][C-0000bd28] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [May 18 08:47:57] VERBOSE[6508][C-0000bd28] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.72.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK11bac3a3;received=192.168.72.254 From: "D.Melekhov" ;tag=as41717c2d To: ;tag=as5380c836 Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 776907409 776907409 IN IP4 192.168.22.19 s=Asterisk PBX 13.13.1 c=IN IP4 192.168.22.19 t=0 0 m=audio 30412 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [May 18 08:47:57] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:47:57] VERBOSE[6514] asterisk.c: Remote UNIX connection disconnected [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.72.254:5060 ---> ACK sip:6000@192.168.22.19:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK700e8406 Max-Forwards: 70 From: "D.Melekhov" ;tag=as41717c2d To: ;tag=as5380c836 Contact: Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (10 headers 0 lines) --- [May 18 08:47:57] VERBOSE[6508][C-0000bd28] pbx.c: Executing [6000@meetmectx:8] MeetMe("SIP/ast-lud-00002a0e", "6000,TL(10800000:60000)") in new stack [May 18 08:47:57] VERBOSE[6508][C-0000bd28] config.c: Parsing '/etc/asterisk/meetme.conf': Found [May 18 08:47:57] VERBOSE[6508][C-0000bd28] app_meetme.c: Created MeetMe conference 1022 for conference '6000' [May 18 08:47:57] VERBOSE[6508][C-0000bd28] app_meetme.c: Setting conference duration limit to: 10800000ms. [May 18 08:47:57] VERBOSE[6508][C-0000bd28] app_meetme.c: Setting warning time to 60000ms from the conference duration limit. [May 18 08:47:57] VERBOSE[6508][C-0000bd28] file.c: Playing 'conf-onlyperson.alaw' (language 'ru') [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> INVITE sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="6053",realm="asterisk",nonce="665e2444",uri="sip:3550@asterisk.p98.belkam.com",algorithm=MD5,response="0fcedcc7874c35e5c304d24a13d745ed" Contact: "6053" Expires: 240 User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces Content-Type: application/sdp v=0 o=- 301205 301205 IN IP4 192.168.158.20 s=- c=IN IP4 192.168.158.20 t=0 0 m=audio 16418 RTP/AVP 18 0 2 8 9 96 97 98 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (15 headers 18 lines) --- [May 18 08:47:57] VERBOSE[22763][C-0000bd27] chan_sip.c: Ignoring this INVITE request [May 18 08:47:57] VERBOSE[22763] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.158.20:5060: SIP/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927;received=192.168.158.20 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as0bb7c79c Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> ACK sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as0bb7c79c Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="6053",realm="asterisk",nonce="665e2444",uri="sip:3550@asterisk.p98.belkam.com",algorithm=MD5,response="0fcedcc7874c35e5c304d24a13d745ed" Contact: "6053" User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 0 <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:47:57] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> ACK sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as0bb7c79c Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="6053",realm="asterisk",nonce="665e2444",uri="sip:3550@asterisk.p98.belkam.com",algorithm=MD5,response="0fcedcc7874c35e5c304d24a13d745ed" Contact: "6053" User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 0 <-------------> [May 18 08:47:57] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:47:58] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.158.20:5060 ---> ACK sip:3550@asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.158.20:5060;branch=z9hG4bK-c6372927 From: "6053" ;tag=c85a513d1eee2fbo0 To: "Малков П,А," ;tag=as0bb7c79c Call-ID: 4bd677e5-d5ccac73@192.168.158.20 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="6053",realm="asterisk",nonce="665e2444",uri="sip:3550@asterisk.p98.belkam.com",algorithm=MD5,response="0fcedcc7874c35e5c304d24a13d745ed" Contact: "6053" User-Agent: Cisco/SPA502G-7.5.2 Content-Length: 0 <-------------> [May 18 08:47:58] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:47:59] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:47:59] VERBOSE[6569] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:01] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:01] VERBOSE[6622] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:01] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.72.254:5060 ---> BYE sip:6000@192.168.22.19:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK5d603dc1 Max-Forwards: 70 From: "D.Melekhov" ;tag=as41717c2d To: ;tag=as5380c836 Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 13.13.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> [May 18 08:48:01] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:48:01] VERBOSE[22763][C-0000bd28] chan_sip.c: Sending to 192.168.72.254:5060 (no NAT) [May 18 08:48:01] VERBOSE[22763][C-0000bd28] chan_sip.c: Scheduling destruction of SIP dialog '1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060' in 32000 ms (Method: BYE) [May 18 08:48:01] VERBOSE[22763][C-0000bd28] chan_sip.c: <--- Transmitting (no NAT) to 192.168.72.254:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.72.254:5060;branch=z9hG4bK5d603dc1;received=192.168.72.254 From: "D.Melekhov" ;tag=as41717c2d To: ;tag=as5380c836 Call-ID: 1eeec4c829b5455f68a858cd3e6bb6f3@192.168.72.254:5060 CSeq: 103 BYE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:48:01] VERBOSE[6508][C-0000bd28] chan_dahdi.c: Hungup 'DAHDI/pseudo-685213114' [May 18 08:48:01] VERBOSE[6508][C-0000bd28] pbx.c: Spawn extension (meetmectx, 6000, 8) exited non-zero on 'SIP/ast-lud-00002a0e' [May 18 08:48:02] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-a12febec From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: 3adf49c-6525cefe@192.168.105.20 CSeq: 1001 SUBSCRIBE Max-Forwards: 70 Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:48:02] VERBOSE[22763] chan_sip.c: --- (12 headers 0 lines) --- [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:02] VERBOSE[22763] sip/route.c: sip_route_dump: route/path hop: [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:48:02] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-a12febec;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as3aa11c8c Call-ID: 3adf49c-6525cefe@192.168.105.20 CSeq: 1001 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6b7d77a4" Content-Length: 0 <------------> [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog '3adf49c-6525cefe@192.168.105.20' in 6400 ms (Method: SUBSCRIBE) [May 18 08:48:02] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-1bb11cc4 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: 3adf49c-6525cefe@192.168.105.20 CSeq: 1002 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="6051",realm="asterisk",nonce="6b7d77a4",uri="sip:asterisk.p98.belkam.com",algorithm=MD5,response="7b4a6b1ec6b2d6ac7cc149ed6b05cb75" Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:48:02] VERBOSE[22763] chan_sip.c: --- (13 headers 0 lines) --- [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:48:02] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-1bb11cc4;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as3aa11c8c Call-ID: 3adf49c-6525cefe@192.168.105.20 CSeq: 1002 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:48:02] VERBOSE[22763] chan_sip.c: Really destroying SIP dialog '3adf49c-6525cefe@192.168.105.20' Method: SUBSCRIBE [May 18 08:48:03] VERBOSE[22763] chan_sip.c: Really destroying SIP dialog '4bd677e5-d5ccac73@192.168.158.20' Method: ACK [May 18 08:48:03] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:03] VERBOSE[6664] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:04] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.28.14:59105 ---> <-------------> [May 18 08:48:04] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:04] VERBOSE[6689] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:05] VERBOSE[5643] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:08] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.26.116:5060 ---> NOTIFY sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.26.116:5060;branch=z9hG4bK-3fa38a2b From: "6054" ;tag=69972247ad9fa1c3o0 To: Call-ID: 79bb9717-f8113333@192.168.26.116 CSeq: 5917 NOTIFY Max-Forwards: 70 Contact: "6054" Event: keep-alive User-Agent: Linksys/SPA922-6.1.5(a) Content-Length: 0 <-------------> [May 18 08:48:08] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:48:08] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.26.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.116:5060;branch=z9hG4bK-3fa38a2b;received=192.168.26.116 From: "6054" ;tag=69972247ad9fa1c3o0 To: ;tag=as43ab47f2 Call-ID: 79bb9717-f8113333@192.168.26.116 CSeq: 5917 NOTIFY Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:48:08] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog '79bb9717-f8113333@192.168.26.116' in 32000 ms (Method: NOTIFY) [May 18 08:48:12] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-bd61ea23 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: 578c88e7-b9fdd320@192.168.105.20 CSeq: 1001 SUBSCRIBE Max-Forwards: 70 Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:48:12] VERBOSE[22763] chan_sip.c: --- (12 headers 0 lines) --- [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:12] VERBOSE[22763] sip/route.c: sip_route_dump: route/path hop: [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:48:12] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-bd61ea23;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as6421a864 Call-ID: 578c88e7-b9fdd320@192.168.105.20 CSeq: 1001 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7b3264ed" Content-Length: 0 <------------> [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog '578c88e7-b9fdd320@192.168.105.20' in 6400 ms (Method: SUBSCRIBE) [May 18 08:48:12] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-7e20c7c7 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: 578c88e7-b9fdd320@192.168.105.20 CSeq: 1002 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="6051",realm="asterisk",nonce="7b3264ed",uri="sip:asterisk.p98.belkam.com",algorithm=MD5,response="0c21d0457def0bd41e776f99fa1a2e27" Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:48:12] VERBOSE[22763] chan_sip.c: --- (13 headers 0 lines) --- [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:48:12] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-7e20c7c7;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as6421a864 Call-ID: 578c88e7-b9fdd320@192.168.105.20 CSeq: 1002 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:48:12] VERBOSE[22763] chan_sip.c: Really destroying SIP dialog '578c88e7-b9fdd320@192.168.105.20' Method: SUBSCRIBE [May 18 08:48:16] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:16] VERBOSE[6914] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:17] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:17] VERBOSE[6940] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:18] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:18] VERBOSE[6963] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:18] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:18] VERBOSE[6971] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:19] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.28.14:59105 ---> <-------------> [May 18 08:48:20] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:20] VERBOSE[7004] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:21] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:21] VERBOSE[7029] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:22] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-82e37431 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: 696ba5b1-e3af50cf@192.168.105.20 CSeq: 1001 SUBSCRIBE Max-Forwards: 70 Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:48:22] VERBOSE[22763] chan_sip.c: --- (12 headers 0 lines) --- [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:22] VERBOSE[22763] sip/route.c: sip_route_dump: route/path hop: [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:48:22] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-82e37431;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as7148dce2 Call-ID: 696ba5b1-e3af50cf@192.168.105.20 CSeq: 1001 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6705d881" Content-Length: 0 <------------> [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog '696ba5b1-e3af50cf@192.168.105.20' in 6400 ms (Method: SUBSCRIBE) [May 18 08:48:22] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.105.20:5060 ---> SUBSCRIBE sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-811ab3b9 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" Call-ID: 696ba5b1-e3af50cf@192.168.105.20 CSeq: 1002 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="6051",realm="asterisk",nonce="6705d881",uri="sip:asterisk.p98.belkam.com",algorithm=MD5,response="81c91dda34e5a6aa557e177cd19926cd" Contact: "6051" Expires: 3600 Event: missed-call-summary User-Agent: Linksys/SPA921-5.1.8 Content-Length: 0 <-------------> [May 18 08:48:22] VERBOSE[22763] chan_sip.c: --- (13 headers 0 lines) --- [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Creating new subscription [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Sending to 192.168.105.20:5060 (no NAT) [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Found peer '6051' for '6051' from 192.168.105.20:5060 [May 18 08:48:22] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.105.20:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK-811ab3b9;received=192.168.105.20 From: "6051" ;tag=3b98ef8464236bc5 To: "6051" ;tag=as7148dce2 Call-ID: 696ba5b1-e3af50cf@192.168.105.20 CSeq: 1002 SUBSCRIBE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:48:22] VERBOSE[22763] chan_sip.c: Really destroying SIP dialog '696ba5b1-e3af50cf@192.168.105.20' Method: SUBSCRIBE [May 18 08:48:23] VERBOSE[22620] asterisk.c: Remote UNIX connection [May 18 08:48:23] VERBOSE[7080] asterisk.c: Remote UNIX connection disconnected [May 18 08:48:23] VERBOSE[22763] chan_sip.c: <--- SIP read from UDP:192.168.26.116:5060 ---> NOTIFY sip:asterisk.p98.belkam.com SIP/2.0 Via: SIP/2.0/UDP 192.168.26.116:5060;branch=z9hG4bK-3e8149d6 From: "6054" ;tag=69972247ad9fa1c3o0 To: Call-ID: 79bb9717-f8113333@192.168.26.116 CSeq: 5918 NOTIFY Max-Forwards: 70 Contact: "6054" Event: keep-alive User-Agent: Linksys/SPA922-6.1.5(a) Content-Length: 0 <-------------> [May 18 08:48:23] VERBOSE[22763] chan_sip.c: --- (11 headers 0 lines) --- [May 18 08:48:23] VERBOSE[22763] chan_sip.c: <--- Transmitting (no NAT) to 192.168.26.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.26.116:5060;branch=z9hG4bK-3e8149d6;received=192.168.26.116 From: "6054" ;tag=69972247ad9fa1c3o0 To: ;tag=as43ab47f2 Call-ID: 79bb9717-f8113333@192.168.26.116 CSeq: 5918 NOTIFY Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [May 18 08:48:23] VERBOSE[22763] chan_sip.c: Scheduling destruction of SIP dialog '79bb9717-f8113333@192.168.26.116' in 32000 ms (Method: NOTIFY)