Регистрация SIP транка у провайдера

MaxSIP · June 17, 2014

Приветствую!

Подскажите где искать косяк, при наборе на выделенный номер, либо занято, либо нет такого номера, видимо чтото с маршрутом что я забываю отправить кроме как авторизоваться.

Есть SIP-UA

sip-ua
credentials username 1хххх3 password 7 1134565445115A555C7E3E realm sip.xxxx.net
authentication username 1xxxx3 password 7 1134565445115A555C7E3E realm sip.xxxx.net
registrar dns:sip.хххх.net expires 3600
sip-server dns:sip.хххх.net

При этом

voip#sh sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
1xxxx3

-1 171 no

Я совершенно видимо не разбираюсь в войс на циско, но по логике вещей мне необходимо отправить некий id провайдеру, чтобы тот сопоставил его и номер, и отправлял все что приходит на номер, на мою циску, некий динамический диалпир,

Поэтому я чтото забываю отправить, верней не чтото, а это самый sip id

В #debug ccsip messages при этом

Jun 17 18:25:00.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.xxxx.net:5060 SIP/2.0
Via: SIP/2.0/UDP xxxx:5060;branch=z9hG4bK831905
From: <sip:xxxx@sip.xxxx.net>;tag=187FFB0-164B
To: <sip:xxxx@sip.xxxx.net>
Date: Tue, 17 Jun 2014 18:25:00 GMT
Call-ID: 1321C387-F57E11E3-8275E7EA-3056A1DF
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1403029500
CSeq: 15 REGISTER
Contact: <sip:xxxx@xxxx:5060>
Expires:  3600
Supported: path
Content-Length: 0


Jun 17 18:25:00.154: //417/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Registering...
Via: SIP/2.0/UDP xxxx:5060;received=xxxx;rport=50190;branch=z9hG4bK831905
From: <sip:xxxx@sip.xxxx.net>;tag=187FFB0-164B
To: <sip:xxxx@sip.xxxx.net>
Call-ID: 1321C387-F57E11E3-8275E7EA-3056A1DF
CSeq: 15 REGISTER
Server: OpenSIPS (1.10.1-tls (x86_64/linux))
Content-Length: 0


Jun 17 18:25:00.154: //417/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxxx:5060;received=xxxx;rport=50190;branch=z9hG4bK831905
From: <sip:xxxx@sip.xxxx.net>;tag=187FFB0-164B
To: <sip:xxxx@sip.xxxx.net>;tag=5443baf9c54e4c976efb0b47ee0dbbfa-3550
Call-ID: 1321C387-F57E11E3-8275E7EA-3056A1DF
CSeq: 15 REGISTER
WWW-Authenticate: Digest realm="n1", nonce="53a0881a000126878e52de2774b65b2c45a68334ef27460c", qop="auth"
Server: OpenSIPS (1.10.1-tls (x86_64/linux))
Content-Length: 0

Andrey77777 · June 18, 2014

Может быть это поможет:

voice translation-rule 100

rule 1 /^9/ //

(для телефона назначения. В данном случае - убрать 9)

voice translation-rule 101

rule 1 /^.*/ /1хххх3/

(1хххх3 - это номер, которым закрывать исходящие звонки)

voice translation-profile SIPTrunk

translate calling 101

translate called 100

voice class sip-profiles 1

request INVITE sip-header From modify "<sip:(.*)@(.*)>" "<sip:1хххх3@sip.хххх.net>"

(преобразование, когда исходящие звонки закрываются не числовыми значениями)

Профили прописать в диалпире

MaxSIP · June 18, 2014

Может быть это поможет:

voice translation-rule 100

rule 1 /^9/ //

(для телефона назначения. В данном случае - убрать 9)

voice translation-rule 101

rule 1 /^.*/ /1хххх3/

(1хххх3 - это номер, которым закрывать исходящие звонки)

voice translation-profile SIPTrunk

translate calling 101

translate called 100

voice class sip-profiles 1

request INVITE sip-header From modify "<sip:(.*)@(.*)>" "<sip:1хххх3@sip.хххх.net>"

(преобразование, когда исходящие звонки закрываются не числовыми значениями)

Профили прописать в диалпире

Все оказалось немного по другому

Почему то SIP-UA хочет authentication без realm sip.xxxx.net

Убрал, и все, регистрация появилась.

Еще одно, при исходящем звонке, провайдер ждет ID, поэтому указал в диалпире clid network-number 1хххх3

dial-peer voice 13 voip
destination-pattern 989151212121
session protocol sipv2
session target sip-server
voice-class codec 1
clid network-number 1xxxx3

Теперь вопрос, как отрезать 9 из того что указанно в destination-pattern

И не получается настроить входящий dial-peer, непонятно что отлавливать, в итоге получаю 403 Forbidden

Received:
INVITE sip:1xxxx3@88.221.215.157:5060 SIP/2.0
Record-Route: <sip:88.201.190.2;lr;ftag=8fc5208c4f8bd07358ac2d0717595bbe;did=f31.ee8a87d2>
Via: SIP/2.0/UDP 88.201.190.2:5060;branch=z9hG4bK8aff.efba1be1.0
Via: SIP/2.0/UDP 88.201.190.2:5060;received=88.201.190.8;branch=z9hG4bKdab22b309ca5b62b6c301b6335d39e60;rport=5060
Max-Forwards: 69
From: +74957969999 <sip:+74957969999@88.201.190.2>;tag=8fc5208c4f8bd07358ac2d0717595bbe
To: <sip:*1xxxx3@sip.n1.xxxx.net>
Call-ID: 292de0f6-5c7795ef-9843397-cbbf@88.201.190.207_b2b_1
CSeq: 201 INVITE
Contact: Anonymous <sip:+74957969999@88.201.190.8:5060>
Expires: 300
User-Agent: Sippy B2BUA (xxxx)
cisco-GUID: 948528939-3913064609-863506136-868101092
h323-conf-id: 948528939-3913064609-863506136-868101092
Content-Length: 230
Content-Type: application/sdp

v=0
o=- 310222 0 IN IP4 88.201.190.211
s=Cisco SDP 0
c=IN IP4 88.201.190.2
t=0 0
m=audio 40282 RTP/AVP 8 0 18 99 102 101
a=rtpmap:99 G.729b/8000
a=rtpmap:102 G.729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Jun 18 10:58:21.917: //3553/3889672B3378/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 88.201.190.2:5060;branch=z9hG4bK8aff.efba1be1.0,SIP/2.0/UDP 88.201.190.2:5060;received=88.201.190.8;branch=z9hG4bKdab22b309ca5b62b6c301b6335d39e60;rport=5060
From: +74957969999 <sip:+74957969999@88.201.190.2>;tag=8fc5208c4f8bd07358ac2d0717595bbe
To: <sip:*1xxxx3@sip.n1.xxxx.net>
Date: Wed, 18 Jun 2014 10:58:21 GMT
Call-ID: 292de0f6-5c7795ef-9843397-cbbf@88.201.190.207_b2b_1
CSeq: 201 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Jun 18 10:58:21.921: //3553/3889672B3378/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 88.201.190.2:5060;branch=z9hG4bK8aff.efba1be1.0,SIP/2.0/UDP 88.201.190.2:5060;received=88.201.190.8;branch=z9hG4bKdab22b309ca5b62b6c301b6335d39e60;rport=5060
From: +74957969999 <sip:+74957969999@88.201.190.2>;tag=8fc5208c4f8bd07358ac2d0717595bbe
To: <sip:*1xxxx3@sip.n1.xxxx.net>;tag=51572B4-9D
Date: Wed, 18 Jun 2014 10:58:21 GMT
Call-ID: 292de0f6-5c7795ef-9843397-cbbf@88.201.190.207_b2b_1
CSeq: 201 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0

Andrey77777 · June 19, 2014

voice translation-rule 100

rule 1 /^9/ //

voice translation-rule 101

rule 1 /^.*/ /1xxxx3/

voice translation-profile SIPTrunk

translate calling 101

translate called 100

dial-peer voice 13 voip

translation-profile outgoing SIPTrunk

destination-pattern 9T

incoming called-number 1...

session protocol sipv2

session target sip-server

session transport udp

voice-class codec 1

voice-class sip early-offer forced

dtmf-relay sip-kpml rtp-nte

MaxSIP · June 19, 2014

voice translation-rule 100

rule 1 /^9/ //

voice translation-rule 101

rule 1 /^.*/ /1xxxx3/

voice translation-profile SIPTrunk

translate calling 101

translate called 100

dial-peer voice 13 voip

translation-profile outgoing SIPTrunk

destination-pattern 9T

incoming called-number 1...

session protocol sipv2

session target sip-server

session transport udp

voice-class codec 1

voice-class sip early-offer forced

dtmf-relay sip-kpml rtp-nte

Неа, лыжи не едут.

Рано еще к рулю подходит.

dial-peer voice 3000 voip
translation-profile incoming INCOMING
session protocol sipv2
session target sip-server
incoming called-number 1xxxx3
voice-class codec 1
dtmf-relay rtp-nte
no vad

диалпир входящий звонок находит(esult=Success(0); Incoming Dial-peer=3000 Is Matched), а вот куда его отправить в моем случае, сип аппараты зарегистрированы.

В случае аналоговых портов, все понятно, с удаленным сип сервером, вроде тоже, а вот куда уйдет звонок после как диалпир найден.

При входящих вызовах, deb ccsip call

Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Calling Number=1xxxx3, Called Number=1xxxx3, Peer Info Type=DIALPEER_INFO_S                                                                                        PEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Match Rule=DP_MATCH_DEST; Called Number=1xxxx3
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Dial String=1xxxx3, Expanded String=1xxxx3, Calling Number=1xxxx3T
  Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring=1xxxx3, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
  Result=NO_MATCH(-1)
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Calling Number=+74957969999, Called Number=, Voice-Interface=0x0,
  Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
  Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_ANSWER; Calling Number=+74957969999
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=+74957969999T
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_ORIGINATE; Calling Number=+74957969999
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=+74957969999T
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Result=NO_MATCH(-1) After All Match Rules Attempt
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:exit@6613
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Calling Number=+74957969999, Called Number=, Voice-Interface=0x0,
  Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
  Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_ANSWER; Calling Number=+74957969999
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=+74957969999T
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_ORIGINATE; Calling Number=+74957969999
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=+74957969999T
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
  Result=NO_MATCH(-1) After All Match Rules Attempt
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
Jun 19 07:56:29.341: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:exit@6613
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Calling Number=+74957969999, Called Number=1xxxx3, Voice-Interface=0x0,
  Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
  Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_VIA_URI; URI=sip:88.201.190.2:5060
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_REQUEST_URI; URI=sip:1xxxx3@88.221.215.157:5060
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_TO_URI; URI=sip:*1xxxx3@sip.n1.xxxxx.net
Jun 19 07:56:29.341: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_FROM_URI; URI=sip:+74957969999@88.201.190.2
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Dial String=, Expanded String=, Calling Number=
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=1xxxx3
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:
  Is Incoming=TRUE, Number Expansion=FALSE
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchCore:
  Dial String=1xxxx3, Expanded String=1xxxx3, Calling Number=
  Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/MatchNextPeer:
  Result=Success(0); Incoming Dial-peer=3000 Is Matched
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchPeertype:exit@5985
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerCore:
  Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=3000
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpMatchSafModulePlugin:
  dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
Jun 19 07:56:29.345: //-1/05EF21ECE8F1/DPM/dpAssociateIncomingPeerSPI:exit@6564
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Calling Number=, Called Number=1xxxx3, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  Match Rule=DP_MATCH_DEST; Called Number=1xxxx3
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Dial String=1xxxx3, Expanded String=1xxxx3, Calling Number=
  Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
  Result=-1
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
  No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
  dialstring=1xxxx3, saf_enabled=0, saf_dndb_lookup=1, dp_result=-1
Jun 19 07:56:29.345: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
  Result=NO_MATCH(-1)
Jun 19 07:56:29.353: //45746/05EF21ECE8F1/SIP/Call/sipSPICallInfo:

Edited June 19, 2014 by MaxSIP

MaxSIP · June 19, 2014

Неужто нет спецов, господа?

awsswa · July 4, 2014

спецов навалом - только никто не хочет разбираться за бесплатно.

mameluk · July 5, 2014

MaxSIP ещё актуально?

Не очень понятно куда вы хотите зарулить входящий звонок.

А вообще - любой звонок на cisco бьется на 2 call-lega. Т.е. для входящего звонка Вам нужно два dial-peer.

Одним ловите входящий по incoming called-number .T, вторым по destination-pattern отправляете в VoIP(или куда надо?). Отмечу, чтобы работал VOIP to VOIP на cisco нужно чтобы был элемент CUBE в прошивке.

Проверить очень просто. Если получается выполнить эту команду

voice service voip

allow-connections sip to sip.

Или вы регаете телефоны на CACME(в начале не написано, на чем терминируются конечные телефоны).

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Регистрация SIP транка у провайдера

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