Asterisk + SIP. Не возвращается звонок с Hold'а

Здравствуйте! Застрял на много дней на одной проблеме, буду очень благодарен за любую помощь в ее решении.

Asterisk 1.6.2.0

Звоню с внешнего мобильного номера через SIP транк на Zadarma. Звонок проходит через астериск и уходит на один из экстеншнов (101). Экстеншн принимает звонок через софтфон на удаленном компе за роутером. Тот принимает вызов, все корректно, звук есть. Проблемы начинаются при постановке звонка на холд. При попытке вернуть звонок софтфон просто отключает линию. При этом у звонящего продолжает играть музыка холда, как будто ничего не случилось. Что самое печальное - ни в CLI, ни в SIP дебаге при этом ничего не появляется. То есть вообще ничего. Есть события в момент постановки звонка на холд, есть события, когда звонящий вешает трубку, но конкретно в этот момент - просто ничего. Тестировал на 4х разных софтфонах (3CX, X-Lite, Ekiga, SJ)

Что вижу в логе. В логе CLI никакой доп. информации к логу SIP debug вроде как нет, приложил его для информации. В логе SIP debug видно (если я правильно понимаю), что в момент постановки звонка на холд почему-то экстеншн 101 начинает слать инвайты обратно на мобильный. Респонсы не доходят, как я понимаю, потому что линия на холде, и все в итоге валится с ошибкой: Maximum retries exceeded on transmission 011c3ae642242519054503f13b54a373@194.28.132.225 for seqno 2 (Non-critical Response)

Не понимаю следующего: 0) Это одна и та же проблема, или две разные? Связано ли то, что респонсы от обратных инвайтов во время холда не могут достучаться до экстеншна с тем, что я не могу вернуть звонок с холда? 1) Зачем экстеншн начинает слать обратные инвайты? Это начинается именно в момент постановки звонка на холд. 2) Что я упускаю? В каком логе еще могут быть следы ошибок при попытке снятия звонка с холда? Была мысль, что раз нет никаких следов в момент возврата с холда, то должна быть ошибка в момент постановки звонка на холд - но нет, так все нормально, только смена статуса экстеншна на 'On Hold' и запуск музыки.

Я в тупике, любая свежая идея очень поможет.

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2012.01.22 17:34:43 =~=~=~=~=~=~=~=~=~=~=~= login as: root Access denied root@194.28.132.225's password: Last login: Sun Jan 22 16:21:13 2012 from ppp91-76-230-161.pppoe.mtu-net.ru Welcome to Elastix ---------------------------------------------------- To access your Elastix System, using a separate workstation (PC/MAC/Linux) Open the Internet Browser using the following URL: http://<YOUR-IP-HERE> If you could not get a DHCP IP address please type setup and select "Network configuration" to set up a static IP. ]0;root@vds:~[root@vds ~]# asterisk -r Asterisk 1.6.2.10, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> 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. ========================================================================= Connected to Asterisk 1.6.2.10 currently running on vds (pid = 9255) vds*CLI> sip set debug offn vds*CLI> SIP Debugging enabled vds*CLI>  <--- SIP read from UDP:78.46.95.118:5060 ---> INVITE sip:101@194.28.132.225 SIP/2.0 Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK6cc69ec1;rport Max-Forwards: 70 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225> Contact: <sip:79671372602@78.46.95.118:5060> Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 102 INVITE User-Agent: Zadarma Voip Date: Sun, 22 Jan 2012 13:34:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 357 v=0 o=root 1199756303 1199756303 IN IP4 78.46.95.118 s=Asterisk PBX 1.8.7.2 c=IN IP4 78.46.95.118 t=0 0 m=audio 15640 RTP/AVP 8 18 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> vds*CLI> --- (14 headers 16 lines) --- vds*CLI> Sending to 78.46.95.118 : 5060 (no NAT) vds*CLI> Using INVITE request as basis request - 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 vds*CLI> Found peer 'zadarma' for '79671372602' from 78.46.95.118:5060 vds*CLI> Found RTP audio format 8 vds*CLI> Found RTP audio format 18 vds*CLI> Found RTP audio format 0 vds*CLI> Found RTP audio format 3 vds*CLI> Found RTP audio format 101 vds*CLI> Found audio description format PCMA for ID 8 vds*CLI> Found audio description format G729 for ID 18 vds*CLI> Found audio description format PCMU for ID 0 vds*CLI> Found audio description format GSM for ID 3 vds*CLI> Found audio description format telephone-event for ID 101 vds*CLI> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) vds*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) vds*CLI> Peer audio RTP is at port 78.46.95.118:15640 vds*CLI> Looking for 101 in from-trunk-sip-zadarma (domain 194.28.132.225) vds*CLI> list_route: hop: <sip:79671372602@78.46.95.118:5060> vds*CLI>  <--- Transmitting (no NAT) to 78.46.95.118:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK6cc69ec1;received=78.46.95.118;rport=5060 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225> Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:101@194.28.132.225> Content-Length: 0 <------------> vds*CLI> Audio is at 194.28.132.225 port 32754 vds*CLI> Adding codec 0x4 (ulaw) to SDP vds*CLI> Adding codec 0x2 (gsm) to SDP vds*CLI> Adding codec 0x8 (alaw) to SDP vds*CLI> Adding non-codec 0x1 (telephone-event) to SDP vds*CLI> Reliably Transmitting (NAT) to 91.76.230.161:58429: INVITE sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1 SIP/2.0 Via: SIP/2.0/UDP 194.28.132.225:5060;branch=z9hG4bK145e1f64;rport Max-Forwards: 70 From: "79671372602" <sip:79671372602@194.28.132.225>;tag=as681e0b3f To: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> Contact: <sip:79671372602@194.28.132.225> Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.10 Date: Sun, 22 Jan 2012 13:34:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 628968834 628968834 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- vds*CLI>  <--- Transmitting (no NAT) to 78.46.95.118:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK6cc69ec1;received=78.46.95.118;rport=5060 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225>;tag=as76a2638a Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:101@194.28.132.225> Content-Length: 0 <------------> vds*CLI>  <--- SIP read from UDP:91.76.230.161:58429 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 194.28.132.225:5060;branch=z9hG4bK145e1f64;rport=5060 Contact: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> To: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d From: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 102 INVITE User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 0 vds*CLI>  <-------------> --- (9 headers 0 lines) --- vds*CLI>  <--- Transmitting (no NAT) to 78.46.95.118:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK6cc69ec1;received=78.46.95.118;rport=5060 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225>;tag=as76a2638a Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:101@194.28.132.225> Content-Length: 0 <------------> vds*CLI>  <--- SIP read from UDP:91.76.230.161:58429 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 194.28.132.225:5060;branch=z9hG4bK145e1f64;rport=5060 Contact: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> To: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d From: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 258 v=0 o=3cxVCE 113117235 214728930 IN IP4 91.76.230.161 s=3cxVCE Audio Call c=IN IP4 91.76.230.161 t=0 0 m=audio 40046 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> vds*CLI> --- (12 headers 11 lines) --- vds*CLI> Found RTP audio format 0 vds*CLI> Found RTP audio format 3 vds*CLI> Found RTP audio format 8 vds*CLI> Found RTP audio format 101 vds*CLI> Found audio description format PCMU for ID 0 vds*CLI> Found audio description format GSM for ID 3 vds*CLI> Found audio description format PCMA for ID 8 vds*CLI> Found audio description format telephone-event for ID 101 vds*CLI> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) vds*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) vds*CLI> Peer audio RTP is at port 91.76.230.161:40046 vds*CLI> list_route: hop: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> vds*CLI> set_destination: Parsing <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> for address/port to send to vds*CLI> set_destination: set destination to 91.76.230.161, port 58429 vds*CLI> Transmitting (NAT) to 91.76.230.161:58429: ACK sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1 SIP/2.0 Via: SIP/2.0/UDP 194.28.132.225:5060;branch=z9hG4bK6a2375d9;rport Max-Forwards: 70 From: "79671372602" <sip:79671372602@194.28.132.225>;tag=as681e0b3f To: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d Contact: <sip:79671372602@194.28.132.225> Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.10 Content-Length: 0 --- vds*CLI> Audio is at 194.28.132.225 port 58126 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 78.46.95.118:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK6cc69ec1;received=78.46.95.118;rport=5060 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225>;tag=as76a2638a Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:101@194.28.132.225> Content-Type: application/sdp Content-Length: 288 v=0 o=root 1951652068 1951652068 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 58126 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> vds*CLI>  <--- SIP read from UDP:78.46.95.118:5060 ---> ACK sip:101@194.28.132.225 SIP/2.0 Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK22a9507f;rport Max-Forwards: 70 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225>;tag=as76a2638a Contact: <sip:79671372602@78.46.95.118:5060> Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 102 ACK User-Agent: Zadarma Voip Content-Length: 0 <-------------> vds*CLI> --- (10 headers 0 lines) --- vds*CLI>  <--- SIP read from UDP:91.76.230.161:58429 ---> INVITE sip:79671372602@194.28.132.225 SIP/2.0 Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:101@91.76.230.161;transport=UDP;rinstance=3d706ffe373368c1> To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE A vds*CLI> llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 6.0.20943.0 Content-Length: 408 v=0 o=3cxVCE 113117235 214728931 IN IP4 91.76.230.161 s=3cxVCE Audio Call c=IN IP4 91.76.230.161 t=0 0 m=audio 40046 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly m=video 40004 RTP/AVP 34 c=IN IP4 91.76.230.161 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1; a=sendrecv <-------------> --- (13 headers 18 lines) --- Sending to 91.76.230.161 : 58429 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found video description format H263 for ID 34 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.76.230.161:40046 <--- Transmitting (NAT) to 91.76.230.161:58429 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Length: 0 <------------> Audio is at 194.28.132.225 port 32754 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 91.76.230.161:58429 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 <------------> vds*CLI> Retransmitting #1 (NAT) to 91.76.230.161:58429: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 --- vds*CLI>  <--- SIP read from UDP:91.76.230.161:58429 ---> <-------------> vds*CLI> Retransmitting #2 (NAT) to 91.76.230.161:58429: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 --- vds*CLI> Retransmitting #3 (NAT) to 91.76.230.161:58429: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 --- vds*CLI> Retransmitting #4 (NAT) to 91.76.230.161:58429: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 --- vds*CLI> Really destroying SIP dialog 'NjJmZTcxNzVkZTIxMGEzMzY4YzQ4NjkwNjYzNjBhYzg.' Method: REGISTER vds*CLI> Retransmitting #5 (NAT) to 91.76.230.161:58429: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 --- vds*CLI> Retransmitting #6 (NAT) to 91.76.230.161:58429: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.76.230.161;branch=z9hG4bK-d8754z-6d372412b642987c-1---d8754z-;received=91.76.230.161;rport=58429 From: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>;tag=e446440d To: "79671372602"<sip:79671372602@194.28.132.225>;tag=as681e0b3f Call-ID: 011c3ae642242519054503f13b54a373@194.28.132.225 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:79671372602@194.28.132.225> Content-Type: application/sdp Content-Length: 308 v=0 o=root 628968834 628968835 IN IP4 194.28.132.225 s=Asterisk PBX 1.6.2.10 c=IN IP4 194.28.132.225 t=0 0 m=audio 32754 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly m=video 0 RTP/AVP 34 --- vds*CLI> [Jan 22 16:35:17] WARNING[9266]: chan_sip.c:3782 retrans_pkt: Maximum retries exceeded on transmission 011c3ae642242519054503f13b54a373@194.28.132.225 for seqno 2 (Non-critical Response) -- See doc/sip-retransmit.txt. vds*CLI> Really destroying SIP dialog '471ea9f154a3463b1dffab6238b14e5b@194.28.132.225' Method: REGISTER vds*CLI>  <--- SIP read from UDP:78.46.95.118:5060 ---> BYE sip:101@194.28.132.225 SIP/2.0 Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK20e90e19;rport Max-Forwards: 70 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225>;tag=as76a2638a Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 103 BYE User-Agent: Zadarma Voip X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 78.46.95.118 : 5060 (no NAT) <--- Transmitting (no NAT) to 78.46.95.118:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.46.95.118:5060;branch=z9hG4bK20e90e19;received=78.46.95.118;rport=5060 From: "79671372602" <sip:79671372602@78.46.95.118>;tag=as78b02721 To: <sip:101@194.28.132.225>;tag=as76a2638a Call-ID: 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060 CSeq: 103 BYE Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> vds*CLI> Scheduling destruction of SIP dialog '011c3ae642242519054503f13b54a373@194.28.132.225' in 14016 ms (Method: INVITE) vds*CLI> Really destroying SIP dialog '26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060' Method: BYE vds*CLI> Disconnected from Asterisk server ]0;root@vds:~[root@vds ~]#