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Не могу настроить транк на ростелеком

https://forum.asterisk.ru/viewtopic.php?t=7695

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Re: Не могу настроить транк на ростелеком

Добавлено: 21 июл 2016, 09:32
Vlad1983
снимать трейс нужно на транке, а не на клиенте

Re: Не могу настроить транк на ростелеком

Добавлено: 21 июл 2016, 09:37
ded
Порты открыл наружу 5090 TCP UDP и 10000 - 20000 TCP UDP
Плочему именно 5090? Такие цифры понравились?
Тайна: открыты или закрыты эти порты наружу на маршрутизаторе - не влияет на исходящие вызовы.

Идите по блок-схеме последовательно.

Re: Не могу настроить транк на ростелеком

Добавлено: 21 июл 2016, 12:43
northug
Я уже замучался снял трейс с транка, к кому обращаться - куда копать ?

Код: Выделить всё

srvsip*CLI> sip set debug ip 62.148.237.152
SIP Debugging Enabled for IP: 62.148.237.152
  == Using SIP RTP CoS mark 5
    -- Executing [322932@test:1] NoOp("SIP/902-000001fe", "") in new stack
    -- Executing [322932@test:2] Dial("SIP/902-000001fe", "SIP/u-tel/322932") in                                                                                                                         new stack
  == Using SIP RTP CoS mark 5
Audio is at 15472
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 62.148.237.152:5060:
INVITE sip:322932@hmngn.usi.ru:5060 SIP/2.0
Via: SIP/2.0/UDP 178.46.176.150:9060;branch=z9hG4bK1ea2bcf8;rport
Max-Forwards: 70
From: "PhonerLite" <sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
To: <sip:322932@hmngn.usi.ru:5060>
Contact: <sip:rpn@178.46.176.150:9060>
Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.6.0
Date: Thu, 21 Jul 2016 09:23:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                                                        H, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 1177011412 1177011412 IN IP4 178.46.176.150
s=Asterisk PBX 13.6.0
c=IN IP4 178.46.176.150
t=0 0
m=audio 15472 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/u-tel/322932

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: "PhonerLite"<sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
To: <sip:322932@hmngn.usi.ru:5060>
Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 178.46.176.150:9060;rport=55247;branch=z9hG4bK1ea2bcf8
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 487 LR2 - User not registered on this client
From: "PhonerLite"<sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
To: <sip:322932@hmngn.usi.ru:5060>;tag=237371852
Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 178.46.176.150:9060;rport=55247;branch=z9hG4bK1ea2bcf8
contact: <sip:322932@hmngn.usi.ru:5060;maddr=62.148.237.152>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 62.148.237.152:5060:
ACK sip:322932@hmngn.usi.ru:5060 SIP/2.0
Via: SIP/2.0/UDP 178.46.176.150:9060;branch=z9hG4bK1ea2bcf8;rport
Max-Forwards: 70
From: "PhonerLite" <sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
To: <sip:322932@hmngn.usi.ru:5060>;tag=237371852
Contact: <sip:rpn@178.46.176.150:9060>
Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '1df662103db640d00b7aba9249f52f83@hmngn.usi                                                                                                                        .ru' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '1df662103db640d00b7aba9249f52f83@hmngn.usi                                                                                                                        .ru' in 6400 ms (Method: INVITE)
    -- No one is available to answer at this time (1:0/0/0)
    -- Auto fallthrough, channel 'SIP/902-000001fe' status is 'NOANSWER'
    -- Registered SIP '1000' at 192.168.0.110:5060
Really destroying SIP dialog '1df662103db640d00b7aba9249f52f83@hmngn.usi.ru' Met                                                                                                                        hod: INVITE
srvsip*CLI> sip set debug off
SIP Debugging Disabled
    -- Registered SIP '8000' at 62.105.2.54:5060
[Jul 21 14:25:11] NOTICE[2585]: chan_sip.c:23908 handle_response_peerpoke: Peer '8000' is now Reachable. (173ms / 2000ms)
srvsip*CLI>   == Using SIP RTP CoS mark 5
srvsip*CLI>     -- Executing [322932@test:1] NoOp("SIP/902-000001fe", "") in new stack
srvsip*CLI>     -- Executing [322932@test:2] Dial("SIP/902-000001fe", "SIP/u-tel/322932") in                                                                                                                         new stack
srvsip*CLI>   == Using SIP RTP CoS mark 5
srvsip*CLI> Audio is at 15472
srvsip*CLI> Adding codec alaw to SDP
srvsip*CLI> Adding codec ulaw to SDP
srvsip*CLI> Adding non-codec 0x1 (telephone-event) to SDP
srvsip*CLI> Reliably Transmitting (NAT) to 62.148.237.152:5060:
srvsip*CLI> INVITE sip:322932@hmngn.usi.ru:5060 SIP/2.0
srvsip*CLI> Via: SIP/2.0/UDP 178.46.176.150:9060;branch=z9hG4bK1ea2bcf8;rport
srvsip*CLI> Max-Forwards: 70
srvsip*CLI> From: "PhonerLite" <sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
srvsip*CLI> To: <sip:322932@hmngn.usi.ru:5060>
srvsip*CLI> Contact: <sip:rpn@178.46.176.150:9060>
srvsip*CLI> Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
srvsip*CLI> CSeq: 102 INVITE
srvsip*CLI> User-Agent: Asterisk PBX 13.6.0
srvsip*CLI> Date: Thu, 21 Jul 2016 09:23:48 GMT
srvsip*CLI> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                                                        H, MESSAGE
srvsip*CLI> Supported: replaces, timer
srvsip*CLI> Content-Type: application/sdp
srvsip*CLI> Content-Length: 279
srvsip*CLI>
srvsip*CLI> v=0
srvsip*CLI> o=root 1177011412 1177011412 IN IP4 178.46.176.150
srvsip*CLI> s=Asterisk PBX 13.6.0
srvsip*CLI> c=IN IP4 178.46.176.150
srvsip*CLI> t=0 0
srvsip*CLI> m=audio 15472 RTP/AVP 8 0 101
srvsip*CLI> a=rtpmap:8 PCMA/8000
srvsip*CLI> a=rtpmap:0 PCMU/8000
srvsip*CLI> a=rtpmap:101 telephone-event/8000
srvsip*CLI> a=fmtp:101 0-16
srvsip*CLI> a=ptime:20
srvsip*CLI> a=maxptime:150
srvsip*CLI> a=sendrecv
srvsip*CLI>
srvsip*CLI> ---
srvsip*CLI>     -- Called SIP/u-tel/322932
srvsip*CLI>
srvsip*CLI> <--- SIP read from UDP:62.148.237.152:5060 --->
srvsip*CLI> SIP/2.0 100 Trying
srvsip*CLI> From: "PhonerLite"<sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
srvsip*CLI> To: <sip:322932@hmngn.usi.ru:5060>
srvsip*CLI> Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
srvsip*CLI> CSeq: 102 INVITE
srvsip*CLI> Via: SIP/2.0/UDP 178.46.176.150:9060;rport=55247;branch=z9hG4bK1ea2bcf8
srvsip*CLI> Content-Length: 0
srvsip*CLI>
srvsip*CLI> <------------->
srvsip*CLI> --- (7 headers 0 lines) ---
srvsip*CLI>
srvsip*CLI> <--- SIP read from UDP:62.148.237.152:5060 --->
srvsip*CLI> SIP/2.0 487 LR2 - User not registered on this client
srvsip*CLI> From: "PhonerLite"<sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
srvsip*CLI> To: <sip:322932@hmngn.usi.ru:5060>;tag=237371852
srvsip*CLI> Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
srvsip*CLI> CSeq: 102 INVITE
srvsip*CLI> Via: SIP/2.0/UDP 178.46.176.150:9060;rport=55247;branch=z9hG4bK1ea2bcf8
srvsip*CLI> contact: <sip:322932@hmngn.usi.ru:5060;maddr=62.148.237.152>
srvsip*CLI> supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
srvsip*CLI> Content-Length: 0
srvsip*CLI>
srvsip*CLI> <------------->
srvsip*CLI> --- (9 headers 0 lines) ---
srvsip*CLI> Transmitting (NAT) to 62.148.237.152:5060:
srvsip*CLI> ACK sip:322932@hmngn.usi.ru:5060 SIP/2.0
srvsip*CLI> Via: SIP/2.0/UDP 178.46.176.150:9060;branch=z9hG4bK1ea2bcf8;rport
srvsip*CLI> Max-Forwards: 70
srvsip*CLI> From: "PhonerLite" <sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
srvsip*CLI> To: <sip:322932@hmngn.usi.ru:5060>;tag=237371852
srvsip*CLI> Contact: <sip:rpn@178.46.176.150:9060>
srvsip*CLI> Call-ID: 1df662103db640d00b7aba9249f52f83@hmngn.usi.ru
srvsip*CLI> CSeq: 102 ACK
srvsip*CLI> User-Agent: Asterisk PBX 13.6.0
srvsip*CLI> Content-Length: 0
srvsip*CLI>
srvsip*CLI>
srvsip*CLI> ---
srvsip*CLI> Scheduling destruction of SIP dialog '1df662103db640d00b7aba9249f52f83@hmngn.usi                                                                                                                        .ru' in 6400 ms (Method: INVITE)
srvsip*CLI> Scheduling destruction of SIP dialog '1df662103db640d00b7aba9249f52f83@hmngn.usi                                                                                                                        .ru' in 6400 ms (Method: INVITE)
srvsip*CLI>     -- No one is available to answer at this time (1:0/0/0)
srvsip*CLI>     -- Auto fallthrough, channel 'SIP/902-000001fe' status is 'NOANSWER'
srvsip*CLI>     -- Registered SIP '1000' at 192.168.0.110:5060
srvsip*CLI> Really destroying SIP dialog '1df662103db640d00b7aba9249f52f83@hmngn.usi.ru' Met

Re: Не могу настроить транк на ростелеком

Добавлено: 21 июл 2016, 12:54
Vlad1983
"SIP/2.0 487 LR2 - User not registered on this client"
узнавайте что они хотят этим сказать

Re: Не могу настроить транк на ростелеком

Добавлено: 21 июл 2016, 15:51
awsswa
Все придумано до нас
http://awsswa.livejournal.com/26586.html

Re: Не могу настроить транк на ростелеком

Добавлено: 22 июл 2016, 00:26
ded
ТС рвёт мозг местному оператору передавая буквы rpn вместо ему разрешённого легального Caller ID
From: "PhonerLite"<sip:rpn@hmngn.usi.ru:9060>;tag=as3cd5a1b3
To: <sip:322932@hmngn.usi.ru:5060>
ну и не факт, что hmngn.usi.ru понимает 6-ти значные номера.

Re: Не могу настроить транк на ростелеком

Добавлено: 22 июл 2016, 09:26
northug
awsswa, ded, Vlad1983 Огромное спасибо за помощь!

Добился !

Код: Выделить всё

Via: SIP/2.0/UDP 178.46.176.150:9060;rport=9060;branch=z9hG4bK17dc359a
Вместо

Код: Выделить всё

Via: SIP/2.0/UDP 178.46.176.150:9060;rport=50208;branch=z9hG4bK5e133741
Часовой пояс: UTC+03:00
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