asterisk.ru

Не идут вызова во статике провайдер МТТ.


Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 80.75.130.136:5060:
INVITE sip:79058896170@80.75.130.136:5060 SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;branch=z9hG4bK2c9d2550
Max-Forwards: 70
From: "73537340300" <sip:73537340300@IP.IP.IP.IP>;tag=as2e215130
To: <sip:79058896170@80.75.130.136:5060>
Contact: <sip:73537340300@IP.IP.IP.IP:5060>
Call-ID: 7c77f3523e0b7e0f0f65e1f1492b69e8@IP.IP.IP.IP:5060
CSeq: 102 INVITE
User-Agent: Linksys/PAP2T-3.1.15(LS)
Date: Thu, 17 Jan 2013 11:06:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 1649577897 1649577897 IN IP4 IP.IP.IP.IP
s=Linksys/PAP2T-3.1.15(LS)
c=IN IP4 IP.IP.IP.IP
t=0 0
m=audio 14402 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;received=89.237.38.33;branch=z9hG4bK2c9d2550;rport=8779
From: "73537340300" <sip:73537340300@IP.IP.IP.IP>;tag=as2e215130
To: <sip:79058896170@80.75.130.136:5060>
Call-ID: 7c77f3523e0b7e0f0f65e1f1492b69e8@IP.IP.IP.IP:5060
CSeq: 102 INVITE

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

<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;received=89.237.38.33;branch=z9hG4bK2c9d2550;rport=8779
From: "73537340300" <sip:73537340300@IP.IP.IP.IP>;tag=as2e215130
To: <sip:79058896170@80.75.130.136:5060>;tag=3864926203-3792803168-380373160-4943405
Call-ID: 7c77f3523e0b7e0f0f65e1f1492b69e8@IP.IP.IP.IP:5060
CSeq: 102 INVITE
Contact: <sip:79058896170@10.133.29.18:5061>
Server: TS-v4.5.1-05c
Reason: MVTSPRO;cause=22;text="Received AccessReject For AccessRequest"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 80.75.130.136:5060:
ACK sip:79058896170@80.75.130.136:5060 SIP/2.0
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;branch=z9hG4bK2c9d2550
Max-Forwards: 70
From: "73537340300" <sip:73537340300@IP.IP.IP.IP>;tag=as2e215130
To: <sip:79058896170@80.75.130.136:5060>;tag=3864926203-3792803168-380373160-4943405
Contact: <sip:73537340300@IP.IP.IP.IP:5060>
Call-ID: 7c77f3523e0b7e0f0f65e1f1492b69e8@IP.IP.IP.IP:5060
CSeq: 102 ACK
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


---
[Jan 17 17:06:45] WARNING[4688]: chan_sip.c:19568 handle_response_invite: Received response: "Forbidden" from '"73537340300" <sip:73537340300@IP.IP.IP.IP>;tag=as2e215130'
[Jan 17 17:06:45] WARNING[4688]: chan_sip.c:19568 handle_response_invite: Received response: "Forbidden" from '"73537340300" <sip:73537340300@IP.IP.IP.IP>;tag=as2e215130'
[Jan 17 17:06:45] WARNING[4819]: app_macro.c:304 _macro_exec: No such context 'macro-hungupcall' for macro 'hungupcall'
[Jan 17 17:06:45] WARNING[4819]: app_macro.c:304 _macro_exec: No such context 'macro-hungupcall' for macro 'hungupcall'
Really destroying SIP dialog '7c77f3523e0b7e0f0f65e1f1492b69e8@IP.IP.IP.IP:5060' Method: INVITE

Дебаг выше, подскажите как решить...

в саппорт МТТ
пусть отрабатываю свою зарплату

да я обратился, а мне в ответ что от меня даже INVATE не приходит до них, причем traceroute проходит отлично

отправьте дебаг на helpdesk@mtt.ru

ТыДыщЬ! (с) ded
http://forum.asterisk.ru/download/file. ... &mode=view

SilentNess писал(а):от меня даже INVATE не приходит до них,

?
Ставьте ngrep. Потом Попутно проясните публике как вы стыкуетесь с МТТ: авторизация по IP или регистрирутесь с А-номером?