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.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: SIP Debug
=~=~=~=~=~=~=~=~=~=~=~= 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.
=========================================================================
[0;37m[0mConnected to Asterisk 1.6.2.10 currently running on vds (pid = 9255)
vds*CLI> sip set debug offn[K
vds*CLI>
[0KSIP Debugging enabled
[Kvds*CLI>
[0K
<--- 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
<------------->
[Kvds*CLI>
[0K--- (14 headers 16 lines) ---
[Kvds*CLI>
[0KSending to 78.46.95.118 : 5060 (no NAT)
[Kvds*CLI>
[0KUsing INVITE request as basis request - 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060
[Kvds*CLI>
[0KFound peer 'zadarma' for '79671372602' from 78.46.95.118:5060
[Kvds*CLI>
[0KFound RTP audio format 8
[Kvds*CLI>
[0KFound RTP audio format 18
[Kvds*CLI>
[0KFound RTP audio format 0
[Kvds*CLI>
[0KFound RTP audio format 3
[Kvds*CLI>
[0KFound RTP audio format 101
[Kvds*CLI>
[0KFound audio description format PCMA for ID 8
[Kvds*CLI>
[0KFound audio description format G729 for ID 18
[Kvds*CLI>
[0KFound audio description format PCMU for ID 0
[Kvds*CLI>
[0KFound audio description format GSM for ID 3
[Kvds*CLI>
[0KFound audio description format telephone-event for ID 101
[Kvds*CLI>
[0KCapabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Kvds*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kvds*CLI>
[0KPeer audio RTP is at port 78.46.95.118:15640
[Kvds*CLI>
[0KLooking for 101 in from-trunk-sip-zadarma (domain 194.28.132.225)
[Kvds*CLI>
[0Klist_route: hop: <sip:79671372602@78.46.95.118:5060>
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0KAudio is at 194.28.132.225 port 32754
[Kvds*CLI>
[0KAdding codec 0x4 (ulaw) to SDP
[Kvds*CLI>
[0KAdding codec 0x2 (gsm) to SDP
[Kvds*CLI>
[0KAdding codec 0x8 (alaw) to SDP
[Kvds*CLI>
[0KAdding non-codec 0x1 (telephone-event) to SDP
[Kvds*CLI>
[0KReliably 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
---
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0K
<--- 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
[Kvds*CLI>
[0K
<------------->
--- (9 headers 0 lines) ---
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0K
<--- 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
<------------->
[Kvds*CLI>
[0K--- (12 headers 11 lines) ---
[Kvds*CLI>
[0KFound RTP audio format 0
[Kvds*CLI>
[0KFound RTP audio format 3
[Kvds*CLI>
[0KFound RTP audio format 8
[Kvds*CLI>
[0KFound RTP audio format 101
[Kvds*CLI>
[0KFound audio description format PCMU for ID 0
[Kvds*CLI>
[0KFound audio description format GSM for ID 3
[Kvds*CLI>
[0KFound audio description format PCMA for ID 8
[Kvds*CLI>
[0KFound audio description format telephone-event for ID 101
[Kvds*CLI>
[0KCapabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Kvds*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kvds*CLI>
[0KPeer audio RTP is at port 91.76.230.161:40046
[Kvds*CLI>
[0Klist_route: hop: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>
[Kvds*CLI>
[0Kset_destination: Parsing <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> for address/port to send to
[Kvds*CLI>
[0Kset_destination: set destination to 91.76.230.161, port 58429
[Kvds*CLI>
[0KTransmitting (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
---
[Kvds*CLI>
[0KAudio 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
<------------>
[Kvds*CLI>
[0K
<--- 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
<------------->
[Kvds*CLI>
[0K--- (10 headers 0 lines) ---
[Kvds*CLI>
[0K
<--- 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
[Kvds*CLI>
[0Kllow: 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
<------------>
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0K
<--- SIP read from UDP:91.76.230.161:58429 --->
<------------->
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KReally destroying SIP dialog 'NjJmZTcxNzVkZTIxMGEzMzY4YzQ4NjkwNjYzNjBhYzg.' Method: REGISTER
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0K[Jan 22 16:35:17] [1;31mWARNING[0m[9266]: [1;37mchan_sip.c[0m:[1;37m3782[0m [1;37mretrans_pkt[0m: Maximum retries exceeded on transmission 011c3ae642242519054503f13b54a373@194.28.132.225 for seqno 2 (Non-critical Response) -- See doc/sip-retransmit.txt.
[Kvds*CLI>
[0KReally destroying SIP dialog '471ea9f154a3463b1dffab6238b14e5b@194.28.132.225' Method: REGISTER
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0KScheduling destruction of SIP dialog '011c3ae642242519054503f13b54a373@194.28.132.225' in 14016 ms (Method: INVITE)
[Kvds*CLI>
[0KReally destroying SIP dialog '26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060' Method: BYE
[Kvds*CLI>
Disconnected from Asterisk server
[0m]0;root@vds:~[root@vds ~]#
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.
=========================================================================
[0;37m[0mConnected to Asterisk 1.6.2.10 currently running on vds (pid = 9255)
vds*CLI> sip set debug offn[K
vds*CLI>
[0KSIP Debugging enabled
[Kvds*CLI>
[0K
<--- 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
<------------->
[Kvds*CLI>
[0K--- (14 headers 16 lines) ---
[Kvds*CLI>
[0KSending to 78.46.95.118 : 5060 (no NAT)
[Kvds*CLI>
[0KUsing INVITE request as basis request - 26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060
[Kvds*CLI>
[0KFound peer 'zadarma' for '79671372602' from 78.46.95.118:5060
[Kvds*CLI>
[0KFound RTP audio format 8
[Kvds*CLI>
[0KFound RTP audio format 18
[Kvds*CLI>
[0KFound RTP audio format 0
[Kvds*CLI>
[0KFound RTP audio format 3
[Kvds*CLI>
[0KFound RTP audio format 101
[Kvds*CLI>
[0KFound audio description format PCMA for ID 8
[Kvds*CLI>
[0KFound audio description format G729 for ID 18
[Kvds*CLI>
[0KFound audio description format PCMU for ID 0
[Kvds*CLI>
[0KFound audio description format GSM for ID 3
[Kvds*CLI>
[0KFound audio description format telephone-event for ID 101
[Kvds*CLI>
[0KCapabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Kvds*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kvds*CLI>
[0KPeer audio RTP is at port 78.46.95.118:15640
[Kvds*CLI>
[0KLooking for 101 in from-trunk-sip-zadarma (domain 194.28.132.225)
[Kvds*CLI>
[0Klist_route: hop: <sip:79671372602@78.46.95.118:5060>
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0KAudio is at 194.28.132.225 port 32754
[Kvds*CLI>
[0KAdding codec 0x4 (ulaw) to SDP
[Kvds*CLI>
[0KAdding codec 0x2 (gsm) to SDP
[Kvds*CLI>
[0KAdding codec 0x8 (alaw) to SDP
[Kvds*CLI>
[0KAdding non-codec 0x1 (telephone-event) to SDP
[Kvds*CLI>
[0KReliably 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
---
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0K
<--- 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
[Kvds*CLI>
[0K
<------------->
--- (9 headers 0 lines) ---
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0K
<--- 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
<------------->
[Kvds*CLI>
[0K--- (12 headers 11 lines) ---
[Kvds*CLI>
[0KFound RTP audio format 0
[Kvds*CLI>
[0KFound RTP audio format 3
[Kvds*CLI>
[0KFound RTP audio format 8
[Kvds*CLI>
[0KFound RTP audio format 101
[Kvds*CLI>
[0KFound audio description format PCMU for ID 0
[Kvds*CLI>
[0KFound audio description format GSM for ID 3
[Kvds*CLI>
[0KFound audio description format PCMA for ID 8
[Kvds*CLI>
[0KFound audio description format telephone-event for ID 101
[Kvds*CLI>
[0KCapabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Kvds*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kvds*CLI>
[0KPeer audio RTP is at port 91.76.230.161:40046
[Kvds*CLI>
[0Klist_route: hop: <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1>
[Kvds*CLI>
[0Kset_destination: Parsing <sip:101@91.76.230.161:58429;transport=UDP;rinstance=3d706ffe373368c1> for address/port to send to
[Kvds*CLI>
[0Kset_destination: set destination to 91.76.230.161, port 58429
[Kvds*CLI>
[0KTransmitting (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
---
[Kvds*CLI>
[0KAudio 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
<------------>
[Kvds*CLI>
[0K
<--- 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
<------------->
[Kvds*CLI>
[0K--- (10 headers 0 lines) ---
[Kvds*CLI>
[0K
<--- 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
[Kvds*CLI>
[0Kllow: 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
<------------>
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0K
<--- SIP read from UDP:91.76.230.161:58429 --->
<------------->
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KReally destroying SIP dialog 'NjJmZTcxNzVkZTIxMGEzMzY4YzQ4NjkwNjYzNjBhYzg.' Method: REGISTER
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0KRetransmitting #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
---
[Kvds*CLI>
[0K[Jan 22 16:35:17] [1;31mWARNING[0m[9266]: [1;37mchan_sip.c[0m:[1;37m3782[0m [1;37mretrans_pkt[0m: Maximum retries exceeded on transmission 011c3ae642242519054503f13b54a373@194.28.132.225 for seqno 2 (Non-critical Response) -- See doc/sip-retransmit.txt.
[Kvds*CLI>
[0KReally destroying SIP dialog '471ea9f154a3463b1dffab6238b14e5b@194.28.132.225' Method: REGISTER
[Kvds*CLI>
[0K
<--- 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
<------------>
[Kvds*CLI>
[0KScheduling destruction of SIP dialog '011c3ae642242519054503f13b54a373@194.28.132.225' in 14016 ms (Method: INVITE)
[Kvds*CLI>
[0KReally destroying SIP dialog '26c5c4607f1549dd26b043a30662296e@78.46.95.118:5060' Method: BYE
[Kvds*CLI>
Disconnected from Asterisk server
[0m]0;root@vds:~[root@vds ~]#
