asterisk.ru

Всем доброго дня.
Читал на данном форуме похожую тему с данной ошибкой, но у меня ситуация чуть другая.
asterisk 12. Провайдер дает номер по sip.
Звонки не ходят только на номера местных мобильных операторов ( сервер в Курске).
Т.е звонок оттуда мне на мобильный или городской номер проходят все отлично.
Если оттуда на городской звонить номер, то же хорошо.
Но вот если звонить на номер местного тел оператора то приходит отлуп. С правильным caller id тему находил на данном форуме, но провайдер то пускает на межгород звонки да и на местную город проходят то же.

Код: Выделить всё

 sip set debug peer 273300
SIP Debugging Enabled for IP: 10.166.100.142
  == Using SIP RTP CoS mark 5
    -- Executing [89508773132@gorod:1] NoOp("SIP/53-000003fd", "") in new stack
    -- Executing [89508773132@gorod:2] Dial("SIP/53-000003fd", "SIP/273300/89508773132") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11828
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.166.100.142:5060:
INVITE sip:89508773132@10.166.100.142 SIP/2.0
Via: SIP/2.0/UDP 10.20.31.84:5060;branch=z9hG4bK76747405
Max-Forwards: 70
From: "T" <sip:273300@10.20.31.84>;tag=as65c87fc4
To: <sip:89508773132@10.166.100.142>
Contact: <sip:273300@10.20.31.84:5060>
Call-ID: 20eee572314aa0f0483e0f00074f46b3@10.20.31.84:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.2.0
Date: Tue, 27 Oct 2015 09:52:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 696102706 696102706 IN IP4 10.20.31.84
s=Asterisk PBX 12.2.0
c=IN IP4 10.20.31.84
t=0 0
m=audio 11828 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/273300/89508773132

<--- SIP read from UDP:10.166.100.142:5060 --->
SIP/2.0 100 Trying
Call-ID: 20eee572314aa0f0483e0f00074f46b3@10.20.31.84:5060
CSeq: 102 INVITE
From: "T" <sip:273300@10.20.31.84>;tag=as65c87fc4
To: <sip:89508773132@10.166.100.142>
Via: SIP/2.0/UDP 10.20.31.84:5060;branch=z9hG4bK76747405
Content-Length: 0

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

<--- SIP read from UDP:10.166.100.142:5060 --->
SIP/2.0 183 Session Progress
Allow: INVITE,ACK,CANCEL,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,PUBLISH
Call-ID: 20eee572314aa0f0483e0f00074f46b3@10.20.31.84:5060
Contact: <sip:89508773132@10.166.100.142:5060>
CSeq: 102 INVITE
From: "T" <sip:273300@10.20.31.84>;tag=as65c87fc4
To: <sip:89508773132@10.166.100.142>;tag=or85rutera
Via: SIP/2.0/UDP 10.20.31.84:5060;branch=z9hG4bK76747405
Content-Length: 163
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=- 8401883 6457234 IN IP4 10.166.100.247
s=-
c=IN IP4 10.166.100.247
b=AS:82
t=0 0
m=audio 22288 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
list_route: route/path hop: <sip:89508773132@10.166.100.142:5060>
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.166.100.247:22288
    -- SIP/273300-000003fe is making progress passing it to SIP/53-000003fd

<--- SIP read from UDP:10.166.100.142:5060 --->
SIP/2.0 500 Server Internal Error
Call-ID: 20eee572314aa0f0483e0f00074f46b3@10.20.31.84:5060
CSeq: 102 INVITE
From: "T" <sip:273300@10.20.31.84>;tag=as65c87fc4
To: <sip:89508773132@10.166.100.142>;tag=or85rutera
Via: SIP/2.0/UDP 10.20.31.84:5060;branch=z9hG4bK76747405
Reason: Q.850;cause=53;text="Outgoing calls barred within CUG"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from 10.166.100.142:5060
set_destination: Parsing <sip:89508773132@10.166.100.142:5060> for address/port to send to
set_destination: set destination to 10.166.100.142:5060
Transmitting (no NAT) to 10.166.100.142:5060:
ACK sip:89508773132@10.166.100.142:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.31.84:5060;branch=z9hG4bK76747405
Max-Forwards: 70
From: "T" <sip:273300@10.20.31.84>;tag=as65c87fc4
To: <sip:89508773132@10.166.100.142>;tag=or85rutera
Contact: <sip:273300@10.20.31.84:5060>
Call-ID: 20eee572314aa0f0483e0f00074f46b3@10.20.31.84:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.2.0
Content-Length: 0


---
    -- SIP/273300-000003fe is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [89508773132@allbell:3] Hangup("SIP/53-000003fd", "") in new stack
  == Spawn extension (allbell, 89508773132, 3) exited non-zero on 'SIP/53-000003fd'
Really destroying SIP dialog '20eee572314aa0f0483e0f00074f46b3@10.20.31.84:5060' Method: INVITE

Спасибо кто поможет вразумиться

Спросить что за ошибка у провайдера
Reason: Q.850;cause=53;text="Outgoing calls barred within CUG"

Благодарю за потраченное время.
Буду решать вопрос с провайдером.

asterisk.ru

got sip response 500 internal server

got sip response 500 internal server

Re: got sip response 500 internal server

Re: got sip response 500 internal server