Ошибка совершения звонка yealink forbidden

IP телефония вытеснила аналоговую практически на 100 % с рынка. Лично я не встречал проводной аналоговый телефон наверно уже лет 5, если не больше. У IP телефонии очень много плюсов, от быстрого подключения до многочисленных настроек и голосовых меню. Но есть и один минус, если раньше не работал аналоговый телефон то проблема была 99% случаев где то на линии, требовалась только сообщить о том что телефон не работает и сидеть ждать. А вот с IP телефонией могут возникнуть различные проблемы, которые придется решать непосредственно тому у кого она установлена. Я уже рассказывал про то как решить различные проблемы возникающие при использование IP телефонии и телефонов фирмы Yealink, можете найти данные статьи через поиск, на самые популярные и полезные приведу ссылки немного ниже. В этой статье рассмотрим ошибку регистрации телефона Yealink T21P, она встречается очень часто.

Просмотр истории звонков Yealink T21

Yealink контакты

Меняем мелодию вызова на Yealink

Yealink режим не беспокоить

Настройка сети Yealink

Что делать если IP телефон Yealink не регистрируется

Как понять в чем причина, ну во первых на самом дисплее телефона появляется сообщение «Ошибка регистрации». Для решение данной проблемы нужно зайти в веб интерфейс телефона. Делается это очень просто, если кто не знает то открываем любой браузер и вводим IP адрес телефона. Узнать IP можно нажав на самом телефоне кнопку «OK».

Дальше необходимо пройти авторизацию введя логин и пароль.

По умолчанию это — admin/admin

После того как открылись настройки, выбираем раздел «Аккаунт» и пункт «Аккаунт» и смотрим статус.

Перед тем как изменять настройки телефона да и вообще искать причину как говориться на своей стороне нужно сделать следующие:

Обязательно позвоните своему оператору, возможно проблемы у него.

Если у оператора все работает пробуем перезагрузить телефон.

Если и это не помогло значить делаем сброс телефона и настраиваем все заново. В моем случае это помогала в 100% случаев. Сделать сброс можно зайдя в раздел «Настройки» и выбрав пункт «Обновление ПО».

После того как все настройки вернутся к заводским, настраиваем акканут и сохраняем изменения.

Еще раз повторю, лично мне избавиться от ошибки регистрации помогал только сброс настроек. Если вы знаете другой способ обязательно поделитесь.

Outgoing calls fail — Forbidden — TotalNet — 03-31-2020 05:40 AM

My business is closed due to the national lockdown here in NZ and I’m trying to get my business phones working from my home office and it’s not going so well. Any help for a Yealink newcomer is gratefully appreciated!

Essentially, I believe my ISP/VoIP provider is blocking my outgoing calls from off-network, however, they have said no at 1st level support so I need some ammo to get through to level 4 ideally.

I have a W60B with 1 W56H and a T41S with the DD10 DECT adapter.

The line is registered and receives incoming calls on that line. When I attempt an outgoing call if fails quickly with «Forbidden».

There are other accounts registered on the W60B with a cloud PBX and my home ISP/VoIP working perfectly. I have set up the T41S for this account without the DECT adapter and get the same failure.

At the business office I use a CISCO ATA (SPA112) for this line and have used its settings for the registration on the W60B, with the intention of replacing it with the Yealink setup soon. This ATA device does not work from my home office either.

I have setup a syslog server to trace the call process and can attach here but the key entries I see are:

— a new call is established (call staus update [0]==>[2])
— a packet is successfully sent to the SIP server and a reply received
— a couple more exchanges of packets with the SIP server
— call status changes to 19 (call status update [2]==>[19])
— the call is released <<RELEASE-REASON>> (0xe2)

There is no SIP proxy involved, the connection is UDP only.

I’m sure there are many reasons behind a «forbidden» call failure (it’s often as much about what’s not in the log) so please let me know if there’s another area I can look at, perhaps it’s my network setup (NAT?) which is why the ATA I brought home doesn’t work either.

Here’s a highlight of the syslog output (phone# replace with NN):

Whether making a call or not I also get the occasional DESV<3+error > 792.450.786:get_ip failed ret= -1

I would port my phone number to the cloud PBX but I get an unbeatable package with unlimited free national and mobile calls on 2 lines

Cheers for any assistance

Richard

RE: Outgoing calls fail — Forbidden — complex1 — 03-31-2020 08:14 AM

(03-31-2020 05:40 AM)TotalNet Wrote:  My business is closed due to the national lockdown here in NZ and I’m trying to get my business phones working from my home office and it’s not going so well. Any help for a Yealink newcomer is gratefully appreciated!

Essentially, I believe my ISP/VoIP provider is blocking my outgoing calls from off-network, however, they have said no at 1st level support so I need some ammo to get through to level 4 ideally.

I have a W60B with 1 W56H and a T41S with the DD10 DECT adapter.

The line is registered and receives incoming calls on that line. When I attempt an outgoing call if fails quickly with «Forbidden».

There are other accounts registered on the W60B with a cloud PBX and my home ISP/VoIP working perfectly. I have set up the T41S for this account without the DECT adapter and get the same failure.

At the business office I use a CISCO ATA (SPA112) for this line and have used its settings for the registration on the W60B, with the intention of replacing it with the Yealink setup soon. This ATA device does not work from my home office either.

I have setup a syslog server to trace the call process and can attach here but the key entries I see are:

— a new call is established (call staus update [0]==>[2])
— a packet is successfully sent to the SIP server and a reply received
— a couple more exchanges of packets with the SIP server
— call status changes to 19 (call status update [2]==>[19])
— the call is released <<RELEASE-REASON>> (0xe2)

There is no SIP proxy involved, the connection is UDP only.

I’m sure there are many reasons behind a «forbidden» call failure (it’s often as much about what’s not in the log) so please let me know if there’s another area I can look at, perhaps it’s my network setup (NAT?) which is why the ATA I brought home doesn’t work either.

Here’s a highlight of the syslog output (phone# replace with NN):

Code:

DCX <5+notice> 632.873.481:TX[0] HS 2 --> T[55597] {CC-INFO}  P_IT(240)
DCX <5+notice> 632.873.625:  <<Calling Party Number>>  (0x6c)
DCX <5+notice> 632.873.769: Number type:  
DCX <5+notice> 632.873.910: Numbering plan: ISDN/telephony plan
DCX <5+notice> 632.874.055: Presentation indicator: , Screening indicator: User-provided, not screened
DCX <5+notice> 632.875.883: Number: 0xNN ("N") 0xNN ("N") 0xNN ("N") 0xNN ("N") 0xNN ("N") 0xNN ("N") 0xNN ("N") 0xNN ("N") 0xNN ("N")  
DCX <5+notice> 632.876.071:  <<Calling Party Name>>  (0x6d)
DCX <5+notice> 632.876.227: "NNNNNNNNN
DCX <5+notice> 632.876.373:  <<Call information>>  (0x7e)
DCX <5+notice> 632.876.518: Call ID:  0x0  
DCX <5+notice> 632.879.676:
DCX <5+notice> 632.879.872:
APP <5+notice> [SIP] SIP_C2S_CALL_NEW_OUTGOING, lid:0, cid:32795,callmask:0x00000000,callee:NNNNNNNNN
CAL <5+notice> [000] call status update [0]==>[2]
DLG <5+notice> [000] Sending Packet :to dest=119.224.142.182:5060 msglen  = 929
DLG <5+notice> [000] End of Sending Packet :msglen  = 929
NET <5+notice> [000] ===>>>> UDP socket 119.224.142.182:5060: send 929 bytes
NET <5+notice> [255] <<<<=== UDP socket 119.224.142.182:5060: read 246 bytes  
DLG <5+notice> [000] Data Received :from src=119.224.142.182:5060 msglen  = 246
DLG <5+notice> [000] End of Data Received :msglen  = 246
NET <5+notice> [255] <<<<=== UDP socket 119.224.142.182:5060: read 277 bytes  
DLG <5+notice> [000] Data Received :from src=119.224.142.182:5060 msglen  = 277
DLG <5+notice> [000] End of Data Received :msglen  = 277
DLG <5+notice> [000] Sending Packet :to dest=119.224.142.182:5060 msglen  = 318
DLG <5+notice> [000] End of Sending Packet :msglen  = 318
NET <5+notice> [000] ===>>>> UDP socket 119.224.142.182:5060: send 318 bytes
CAL <5+notice> [000] call status update [2]==>[19]
SS7 <5+notice> 633.047.590:SS7_ReqRelExt instance=0x1003000e, reason=0x75,msg:(null) rel_msg_len=0
SS7 <5+notice> 633.048.034:service_SS7_ReqRel inst=0x1003000e,Reason=117,Call Id:0
APPC<5+notice> 633.048.266:appmedia_CallObjMediaStop inst:0xf,ChannelID:0
APPC<5+notice> 633.048.453:CallObjMediaSet inst:0xf,state:1
APPC<5+notice> 633.050.701:appcall_ReleaseCall CallId:0, Reason:0x75, RelMsgLen:0, CallIns:0xf, Hs:2
DCX <5+notice> 633.261.418:
DCX <5+notice> 633.261.635:FTMLP_ID/0,3,8,0x0,0x0,0,<0>
DCX <5+notice> 633.291.416:
DCX <5+notice> 633.291.633:FTMLP_ID/0,3,7,0x4,0x0,4,<0>
DCX <5+notice> 633.331.383:
DCX <5+notice> 633.331.590:FTMLP_ID/0,3,7,0x5,0x0,5,<0>
DCX <5+notice> 633.351.965:
DCX <5+notice> 633.352.222:FTMLP_ID/0,3,7,0x6,0x0,6,<0>
DCX <5+notice> 633.352.379:
DCX <5+notice> 633.352.528:RX[0] HS 2 <-- T(55645) {CC-RELEASE-COM} [HS-2]  P_IT(140)
DCX <5+notice> 633.352.676:  <<RELEASE-REASON>>  (0xe2)
DCX <5+notice> 633.352.819: Normal

Whether making a call or not I also get the occasional DESV<3+error > 792.450.786:get_ip failed ret= -1

I would port my phone number to the cloud PBX but I get an unbeatable package with unlimited free national and mobile calls on 2 lines

Cheers for any assistance

Richard

Hi Richard,

Let me sum it up if I understand your issue.
At your business office you use a W60B with two accounts, 1 business and 1 private.
The W56H and T41S are both connected to the W60B.
All work well, no issues.

Because of the lockdown you must work at home, so you take the W60B, W56H and T41S with you and place it in your home office.

Now you have issues… and only with outbound business calls.
No issues with incoming calls, business or private, at all.

I do not think there are issues with you modem/router or other network component at your home, but you can always check if the SIP ALG feature in your modem/router is disabled.
SIP ALG should be a SIP helper but it create more issues than it is doing good.
Also check the port (forward) setting of your business and home router if they are setup different.
Why: SPA112 at the office don’t work… different LAN IP-address and that’s why it cannot communicate with your business VoIP provider?

It is a nasty issue.

RE: Outgoing calls fail — Forbidden — TotalNet — 03-31-2020 11:32 AM

(03-31-2020 08:14 AM)complex1 Wrote:  Let me sum it up if I understand your issue.
At your business office you use a W60B with two accounts, 1 business and 1 private.
The W56H and T41S are both connected to the W60B.
All work well, no issues.

Not quite. The Yealink setup is new and not yet deployed so I am trying to get it working from home by registering the main business phone number on the W60B to make/take business calls at home. The Cisco ATA was working at the office and I have brought it home, I can’t make calls from home on this either. Both register and can receive calls with audio working in both directions, they just can’t initiate calls.

I have 4 other personal SIP accounts with 3 different providers that do work at home on the Yealink setup.

Thanks for the tip on SIP ALG, I have disabled this on the Unifi USG in the controller. This hasn’t made a difference yet but I will reboot everything in the morning to see if that changes. This also gives me something to discuss with tech support at my ISP but I don’t expect much, on a business package they set me up with a consumer grade Router/WiFi/modem/Firewall (that got hacked within 2 days of going online) and if you’re not using that then support kind of halts!

RE: Outgoing calls fail — Forbidden — complex1 — 03-31-2020 12:18 PM

(03-31-2020 11:32 AM)TotalNet Wrote:  The Yealink setup is new and not yet deployed so I am trying to get it working from home by registering the main business phone number on the W60B to make/take business calls at home. The Cisco ATA was working at the office and I have brought it home, I can’t make calls from home on this either. Both register and can receive calls with audio working in both directions, they just can’t initiate calls.

I guess your provider is blocking unknown IP-addresses to avoid unauthorized calls.
They only validated your main office IP-address.

SOLVED: Outgoing calls fail — Forbidden — TotalNet — 05-07-2020 08:30 PM

(03-31-2020 12:18 PM)complex1 Wrote:  

(03-31-2020 11:32 AM)TotalNet Wrote:  The Yealink setup is new and not yet deployed so I am trying to get it working from home by registering the main business phone number on the W60B to make/take business calls at home. The Cisco ATA was working at the office and I have brought it home, I can’t make calls from home on this either. Both register and can receive calls with audio working in both directions, they just can’t initiate calls.

I guess your provider is blocking unknown IP-addresses to avoid unauthorized calls.
They only validated your main office IP-address.

Turns out this is the most likely case, the Yealink phones are now in the office with the only change being the internal IP address and working well.

I was hoping the syslog output of the W60B would include more detail on the response from the SIP proxy like a response code, is that encrypted in the packet received from the proxy or does the proxy even give detail?

There’s no support article here to indicate what call status 19 actual means or if it’s just a generic response for call failure.

My home ISP is part of the same group of companies as the office ISP and tech support said it shouldn’t be a problem. The evidence suggests otherwise — that my home connection is considered off-network.

Thanks for all the replies. Disappointed that my ISP couldn’t assist during an event like this but so far liking the Yealink products.

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK43d6f2c9;rport=60638
From: <sip:380892506511@sip.ukrtel.net>;tag=as4b83d39a
To: <sip:380892506511@195.5.0.83>;tag=aprqcvrsqu2-p2r94110000u6
Call-ID: 7fae610b155ea1e37031b9577cd780e1@127.0.1.1
CSeq: 111 REGISTER
Contact: <sip:380892506511@195.5.0.83>;expires=30

<————->
— (7 headers 0 lines) —
Scheduling destruction of SIP dialog ‘7fae610b155ea1e37031b9577cd780e1@127.0.1.1’ in 32000 ms (Method: REGISTER)
[Jul 30 15:05:31] NOTICE[1767]: chan_sip.c:18399 handle_response_register: Outbound Registration: Expiry for sip.ukrtel.net is 120 sec (Scheduling reregistration in 105 s)
== Using SIP RTP CoS mark 5
— Executing [90800506800@phones:1] Verbose(«SIP/101-00000028», «########## call from the SIP_ALL_OUT ###########») in new stack
########## call from the SIP_ALL_OUT ###########
— Executing [90800506800@phones:2] Dial(«SIP/101-00000028», «SIP/ukrtelecom/0800506800,120») in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.10.10.110 port 15890
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.5.0.83:5060:
INVITE sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK671ecf1e;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>
Contact: <sip:380892506511@10.10.10.110>
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Date: Mon, 30 Jul 2012 12:05:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 1856866922 1856866922 IN IP4 10.10.10.110
s=Asterisk PBX 1.6.2.9-2+squeeze6
c=IN IP4 10.10.10.110
t=0 0
m=audio 15890 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

—
— Called ukrtelecom/0800506800

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK671ecf1e;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 102 INVITE

<————->
— (6 headers 0 lines) —

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 407 Proxy authentication required
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK671ecf1e;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=896FD33D68B66E113617ECF08AC04D4013436499839011779
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 102 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm=»sip.ukrtel.net»,domain=»sip.ukrtel.net»,nonce=»MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==»,algorithm=MD5
Organization: Ukrtelecom

<————->
— (9 headers 0 lines) —
Transmitting (NAT) to 195.5.0.83:5060:
ACK sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK671ecf1e;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=896FD33D68B66E113617ECF08AC04D4013436499839011779
Contact: <sip:380892506511@10.10.10.110>
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Content-Length: 0

—
Audio is at 10.10.10.110 port 15890
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.5.0.83:5060:
INVITE sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK69060425;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>
Contact: <sip:380892506511@10.10.10.110>
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username=»380892506511″, realm=»sip.ukrtel.net», algorithm=MD5, uri=»sip.ukrtel.net», nonce=»MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==», response=»ba9ce3cbc3d573ec0fb047860c6b1091″
Date: Mon, 30 Jul 2012 12:05:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 1856866922 1856866923 IN IP4 10.10.10.110
s=Asterisk PBX 1.6.2.9-2+squeeze6
c=IN IP4 10.10.10.110
t=0 0
m=audio 15890 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

—

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 103 INVITE

<————->
— (6 headers 0 lines) —

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 103 INVITE
Contact: <sip:0800506800@195.5.0.83:5060;transport=udp>
Allow: INVITE
Allow: ACK
Allow: PRACK
Allow: SUBSCRIBE
Allow: BYE
Allow: CANCEL
Allow: NOTIFY
Allow: INFO
Allow: REFER
Allow: UPDATE
Content-Type: application/sdp
Content-Length: 151

v=0
o=- 1 2 IN IP4 195.5.0.83
s=-
c=IN IP4 195.5.0.83
t=0 0
m=audio 20854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<————->
— (19 headers 8 lines) —
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us — 0xc (ulaw|alaw), peer — audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined — 0x8 (alaw)
Non-codec capabilities (dtmf): us — 0x1 (telephone-event), peer — 0x1 (telephone-event), combined — 0x1 (telephone-event)
Peer audio RTP is at port 195.5.0.83:20854
— SIP/ukrtelecom-00000029 is ringing
— SIP/ukrtelecom-00000029 is making progress passing it to SIP/101-00000028

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 103 INVITE
Contact: «Main800506800» <sip:0800506800@195.5.0.83:5060;transport=udp>
Allow: INVITE
Allow: ACK
Allow: PRACK
Allow: SUBSCRIBE
Allow: BYE
Allow: CANCEL
Allow: NOTIFY
Allow: INFO
Allow: REFER
Allow: UPDATE
P-Asserted-Identity: «Main800506800» <sip:380892222600@corp.ukrtelecom.loc:5061>
Content-Type: application/sdp
Content-Length: 151

v=0
o=- 1 2 IN IP4 195.5.0.83
s=-
c=IN IP4 195.5.0.83
t=0 0
m=audio 20854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<————->
— (20 headers 8 lines) —
— SIP/ukrtelecom-00000029 is ringing
— SIP/ukrtelecom-00000029 is making progress passing it to SIP/101-00000028

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 103 INVITE
Contact: «AIC Route to VP» <sip:0800506800@195.5.0.83:5060;transport=udp>
P-Asserted-Identity: «AIC Route to VP» <sip:380892222201@corp.ukrtelecom.loc:5061>
Allow: INVITE
Allow: ACK
Allow: PRACK
Allow: SUBSCRIBE
Allow: BYE
Allow: CANCEL
Allow: NOTIFY
Allow: INFO
Allow: REFER
Allow: UPDATE
Content-Type: application/sdp
Content-Length: 151

v=0
o=- 1 2 IN IP4 195.5.0.83
s=-
c=IN IP4 195.5.0.83
t=0 0
m=audio 20854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<————->
— (20 headers 8 lines) —
list_route: hop: <sip:0800506800@195.5.0.83:5060;transport=udp>
set_destination: Parsing <sip:0800506800@195.5.0.83:5060;transport=udp> for address/port to send to
set_destination: set destination to 195.5.0.83, port 5060
Transmitting (NAT) to 195.5.0.83:5060:
ACK sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK551d3960;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Contact: <sip:380892506511@10.10.10.110>
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Content-Length: 0

—
— SIP/ukrtelecom-00000029 answered SIP/101-00000028

<— SIP read from UDP:195.5.0.83:5060 —>
UPDATE sip:380892506511@10.10.10.110 SIP/2.0
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000g00.1
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 1 UPDATE
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: «101» <sip:380892506511@195.5.0.83>;tag=as6f72688f
Content-Length: 0
Contact: «AIC Route to VP» <sip:0800506800@195.5.0.83:5060;transport=udp>
Max-Forwards: 8
Session-Expires: 1800
Supported: timer
P-Asserted-Identity: «Main800506800» <sip:892506511@10.254.10.17;user=phone>

<————->
— (12 headers 0 lines) —

<— Transmitting (NAT) to 195.5.0.83:5060 —>
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000g00.1;received=195.5.0.83
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: «101» <sip:380892506511@195.5.0.83>;tag=as6f72688f
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 1 UPDATE
Server: Asterisk PBX 1.6.2.9-2+squeeze6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<————>
[Jul 30 15:05:48] NOTICE[1767]: chan_sip.c:22000 handle_incoming: Unknown SIP command ‘UPDATE’ from ‘195.5.0.83’

<— SIP read from UDP:195.5.0.83:5060 —>
UPDATE sip:380892506511@10.10.10.110 SIP/2.0
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000010.1
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 2 UPDATE
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: «101» <sip:380892506511@195.5.0.83>;tag=as6f72688f
Content-Length: 0
Contact: «AIC Route to VP» <sip:0800506800@195.5.0.83:5060;transport=udp>
Max-Forwards: 8
Session-Expires: 1800
Supported: timer
P-Asserted-Identity: «VP0159» <sip:892506511@10.254.10.17;user=phone>

<————->
— (12 headers 0 lines) —

<— Transmitting (NAT) to 195.5.0.83:5060 —>
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000010.1;received=195.5.0.83
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: «101» <sip:380892506511@195.5.0.83>;tag=as6f72688f
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 2 UPDATE
Server: Asterisk PBX 1.6.2.9-2+squeeze6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<————>
[Jul 30 15:05:48] NOTICE[1767]: chan_sip.c:22000 handle_incoming: Unknown SIP command ‘UPDATE’ from ‘195.5.0.83’
Scheduling destruction of SIP dialog ’12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net’ in 6400 ms (Method: UPDATE)
set_destination: Parsing <sip:0800506800@195.5.0.83:5060;transport=udp> for address/port to send to
set_destination: set destination to 195.5.0.83, port 5060
Reliably Transmitting (NAT) to 195.5.0.83:5060:
BYE sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK46c7a376;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username=»380892506511″, realm=»sip.ukrtel.net», algorithm=MD5, uri=»sip.ukrtel.net», nonce=»MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==», response=»49cd1e38a8fe27f4a910af7c1b757cc0″
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

—
== Spawn extension (phones, 90800506800, 2) exited non-zero on ‘SIP/101-00000028’
Retransmitting #1 (NAT) to 195.5.0.83:5060:
BYE sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK46c7a376;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username=»380892506511″, realm=»sip.ukrtel.net», algorithm=MD5, uri=»sip.ukrtel.net», nonce=»MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==», response=»49cd1e38a8fe27f4a910af7c1b757cc0″
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

—
Retransmitting #2 (NAT) to 195.5.0.83:5060:
BYE sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK46c7a376;rport
Max-Forwards: 70
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username=»380892506511″, realm=»sip.ukrtel.net», algorithm=MD5, uri=»sip.ukrtel.net», nonce=»MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==», response=»49cd1e38a8fe27f4a910af7c1b757cc0″
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

—

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK46c7a376;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 104 BYE
Content-Length: 0

<————->
— (7 headers 0 lines) —
Really destroying SIP dialog ’12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net’ Method: UPDATE

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK46c7a376;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 104 BYE
Content-Length: 0

<————->
— (7 headers 0 lines) —

<— SIP read from UDP:195.5.0.83:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK46c7a376;rport=60638
From: «101» <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID:

12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net

CSeq: 104 BYE
Content-Length: 0

<————->
— (7 headers 0 lines) —
energotourserver*CLI>