МТТ и входящие звонки Опции

[foxyfive]

Здравствуйте.
Кто делал подключение к провайдеру МТТ? У них чтобы заработали входящие, нужно прописать sip номер в DID, куда именно, на моей станции, сами не знают. Исходящие звонки работают. С провайдером sipnet у меня проходили входящие.
Снимали дамп входящих, говорят станция не отвечает при входящем.
Буду благодарен за помощь

[foxyfive]

Прокинул порты 5060 и 5061 на ip mfim. Теперь в трассировке выдает:




================================================================================
Received 1270 Bytes from 80.75.130.83:35970 by TCP (ReceiveEv)
--------------------------------------------------------------------------------
INVITE sip:8831407764**262@31.15.**.145:5060;transport=TCP;realip=31.15.**.145 SIP/2.0
Via: SIP/2.0/TCP 80.75.130.83:5060;branch=z9hG4bK5c0r3u109838260svkt0.1
Max-Forwards: 68
Contact: "Anonymous"<sip:795285**367@80.75.130.83:5060;transport=tcp>
To: <sip:8831407764**262@31.15.**.145>
From: <sip:795285**367@80.75.130.83>;tag=zl77rd5nui5clii5.o
Call-ID: 11ac86a8ef8c11e5bbdeac162d8d46a4@10.133.29.22~3o
CSeq: 658 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
P-Asserted-Identity: <sip:795285**367@80.75.132.72>
Remote-Party-ID: <sip:795285**367@80.75.132.72>;party=calling
h323-conf-id: 800184755-3114466536-4282916532-3355231324
Portasip-3264-action: offer 1
cisco-GUID: 800184755-3114466536-4282916532-3355231324
Content-Length: 464

v=0
o=Sippy 1150962848117067538 0 IN IP4 80.75.130.83
s=-
t=0 0
m=audio 56638 RTP/AVP 0 8 18 4 96 97 98 101
c=IN IP4 80.75.130.83
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:96 G729/8000
a=fmtp:96 annexb=no
a=rtpmap:97 G729/8000
a=fmtp:97 annexb=yes
a=rtpmap:98 G723/8000
a=fmtp:98 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

================================================================================

185812-[Sipm_SipTransportMsgReceivedExt]..method(INVITE)..
185812-[Sipm_SipEvCallCreate] SIPCO Call
185812-
[[[[[[[[[[@@@@@ [SIPM] Creation Call ID => 38 (0)@@@@@]]]]]]]]]]
185812-
[[[[[[[[[[@@@@@ [SIPM] Creation Call SS ID => 39 @@@@@]]]]]]]]]]
185812-[CallIdx:38(All:200)][Sipm_SipCallCreate](max:1 OUT:0,IN:1)
185812-[CallIdx:38][Sipm_SipEvCallCreate] INCOMMING call was created
185813-[CallIdx:38][Sipm_SipEvCallMsgReceive] SIPM <--- INVITE
185813-[CallIdx:38][Sipm_SipConnProcInviteInd] reqUriAddr(8831407764**262)
185813-[CallIdx:38][Sipm_SipConnProcInviteInd] fromUser(795285**367)
185813-[CallIdx:38][Sipm_SipConnProcInviteInd] Sim_SipCheckIPForIPAuthen...!!!
185813-[Sipm_SipSaveBlockedIPForIPAuthen] Sun, 21 Mar 2016 20:33:59 IP:80.75.130.83 ID:795285**367 INVITE
185813-[CallIdx:38][Sipm_SipEvCallMsgReceive] Invite Message Failed, so Disconnect...
185813-=============================================
185813-Receive SDP MSG[38] =>
v=0
o=Sippy 1150962848117067538 0 IN IP4 80.75.130.83
s=-
t=0 0
m=audio 56638 RTP/AVP 0 8 18 4 96 97 98 101
c=IN IP4 80.75.130.83
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:96 G729/8000
a=fmtp:96 annexb=no
a=rtpmap:97 G729/8000
a=fmtp:97 annexb=yes
a=rtpmap:98 G723/8000
a=fmtp:98 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
185813-=============================================
185813-[CallIdx:38][Sipm_SipEvCallLegSessionTimerNegotiationFault] - 3
185813-[CallIdx:38][Sipm_SipEvCallState(s:04)] INCOMING - Offering(reason:REMOTE_INVITING)
185813-[CallIdx:38][Sipm_SipEvCallState(s:09)] INCOMING - Terminated(reason:CALL_TERMINATED)
185813-[CallIdx:38][Sipm_SipEvCallState] Terminated
185813-[[[[[[[[[[@@@@@ [SIPM] Delete callId => 38 @@@@@]]]]]]]]]]

185813-========@@@======== SIP Call SS Str[38] : test point 01
185813-[CallIdx:38][Sipm_SipCallInfoDelete](1,0,0)