Sip trunk

[polinoma]

Здравейте,

 

някой конфигурирал ли е Sip trunk към MTEL?

 

Ако някой може да ми даде малко информация .... ще съм благодарен.

 

 

 

[Mihail Peltekov]

Погледни тук

[polinoma]

За asterisk не ми върши работа.

 

Мога да пратя моя конфиг ако някой може да помогне ...

[Mihail Peltekov]

А  написал ли си какво е сиското?

[polinoma]

Здравейте, ползвам Cisco 2811 (като всичко правя под конзола).

 

Пробвах с един друг sip trunk (в БГ) и нямам проблеми, но с МТЕЛ не мога да се оправя.

 

Ще съм благодарен на всеки, който даде някаква насока

 

ето и част от настройките, които имам:

 

 

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol cisco
 h323
 sip
  registrar server expires max 250 min 200

 

 

 

-Тук променям хедърите (посочил съм само част от тях).

 

!
voice class sip-profiles 1
 request INVITE sip-header From modify "<sip:35988XXXXXX@sip.mtel.bg>" "<sip:Username@sip.mtel.bg>"
!

!
voice translation-rule 5
 rule 1 /^.*/ /35988XXXXXX/
!

voice translation-profile SIP.MTEL-Outgoing
 translate calling 5


dial-peer voice 2 voip
 description **Outgoing Calls to sip.mtel.bg SIP Trunk**
 translation-profile outgoing SIP.MTEL-Outgoing
 destination-pattern 0.........
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip profiles 1
 session protocol sipv2
 session target dns:sip.mtel.bg:5060
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
!
sip-ua
 authentication username XXXX password 0  XXXXX realm sip.mtel.bg
 no remote-party-id
 retry invite 2
 retry register 10
 timers connect 100
 registrar dns:register.mtel.bg expires 200
 sip-server dns:sip.mtel.bg
 host-registrar
!

 

 

Ето я и грешките, която получавам:

 

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK7B1791;rport=31345

From: "User" <sip:Username@sip.mtel.bg>;tag=EC9E34-1EBD

To: <sip:088xxxxxx@sip.mtel.bg>

Call-ID: 3BF3E67-B06D11E4-8015AB95-1D2DE580@xxx.xxx.xxx.xxx

CSeq: 101 INVITE

Timestamp: 1423580698

Feb 10 15:04:59.018: //125/0287F73F8010/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX

Feb 10 15:04:59.018: //125/0287F73F8010/SIP/State/sipSPIChangeState: 0x499145B0 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)

Feb 10 15:04:59.018: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length: Could not get Content-length

Feb 10 15:04:59.018: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [213.226.43.205]:5060

Feb 10 15:04:59.018: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1

Feb 10 15:04:59.018: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x486BDE60

Feb 10 15:04:59.018: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x486BDE60, addr=213.226.43.205, port=5060, connid=2, transport=UDP

Feb 10 15:04:59.018: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog

Feb 10 15:04:59.022: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK7B1791;rport=31345

From: "User" <sip:Username@sip.mtel.bg>;tag=EC9E34-1EBD

To: <sip:088XXXXX@sip.mtel.bg>;tag=aprqngfrt-s6t43o30000a6

Call-ID: 3BF3E67-B06D11E4-8015AB95-1D2DE580@xxx.xxx.xxx.xxx

CSeq: 101 INVITE

Timestamp: 1423580698

Reason: Q.850;cause=55;text="Call Terminated"

The Call Setup Information is:

Call Control Block (CCB) : 0x499145B0

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 35988XXXXX

Called Number            : 088xxxxxx

Source IP Address (Sig  ): 1XXXXXXXXX

Destn SIP Req Addr:Port  : 213.226.43.205:5060

Destn SIP Resp Addr:Port : 213.226.43.205:5060

Destination Name         : sip.mtel.bg

Feb 10 15:04:59.030: //125/0287F73F8010/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : No Codec

Negotiated Codec Bytes   : 0

Nego. Codec payload      : 255 (tx), 255 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 1XXXXXXXXX

Source IP Port    (Media): 18798

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

Feb 10 15:04:59.030: //125/0287F73F8010/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 57

Disconnect Cause (SIP)   : 403

 

Странна работа, пробвах и с Asterisk. Виждам, че трънка се регистрира в Мтел.

Набирам номер и следва едно огромно забавяне (около 50-60 сек.) след което чувам как X-lite-a започва да звъни но незнайно къде.

В debug-а виждам, че набира номера, който съм набрал от X-lite-a.

 

Пробвах и с друг trunk и там нещата се получават без абсолютно никакви проблеми.

 

Някакви специални настройки ли имат МТЕЛ?!