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Problem connecting SIP Trunk to ISP

Hi,

I am trying to connect my CME v9.1 (2921) with a SIP Trunk to my ISP.

Right now my connection is like this SCCP Phone --> CME --> FXO --> PSTN, the PSTN is connected using FXO ports, with 8 independent lines, but this lines are connected to 6 DLink Voip Telephone Adapters (DVG-5121SP), which connects to the ISP through a Ethernet RJ45 cable each. (everything works fine).

I would like to remove these VoIP Telephone Adapters, and i am trying to configure the SIP Trunk but it doesn't work.

First of all, I think i don't need CUBE (i don't have the licenses), i hope i am correct.

Second, i am configuring sip-ua, only with one user and password for one line, i am configuring this:

sip-ua

credentials username xxxxx password xxxxx

no remote-party-id

retry invite 2

retry register 10

retry options 0

timers connect 100

registrar dns:172.23.7.134 expires 3600

sip-server dns:172.23.7.134

host-registrar

dial-peer voice 105 voip
description Local Calls
destination-pattern 0[2-7].......

session protocol sipv2
session target ipv4:172.23.7.134
codec g711ulaw

voice translation-rule 3
rule 1 /23823867/ /4000/


voice translation-profile entrantes
translate called 3


dial-peer voice 106 voip
translation-profile incoming entrantes
session protocol sipv2
incoming called-number 23823867
codec g711ulaw
dtmf-relay rtp-nte
no vad

but this doesn't work, i can't dial calls, nor receive them.

how could i test that everything is correct from my side?.

i issued the command debug ccsip messages, and obtain the log i am attaching..

i did this test with a SIP phone (i couldn't make it with a sccp phone, maybe that was the problem), with IP 172.16.164.90, which was registered to my CME (IP: 172.16.161.2), the dn number is 4105, i tried to call 045118106 (using dialpeer 105).

Contact: <sip:4105@172.16.164.90:3077>

To: <sip:045118104@172.16.161.2:5060> Contact: <sip:4105@172.16.164.90:3077>
To: <sip:045118104@172.16.161.2:5060>

hope somebody could help me with this or give me ideas of what i am doing wrong.

regards,

Juan Pablo

5 REPLIES
Hall of Fame Super Gold

Problem connecting SIP Trunk to ISP

You need a translation rule to route calls to extensions.

There are many posts on the subject, use the search funcion.

Problem connecting SIP Trunk to ISP

Hi Paolo,

thanks for your reply,

but i have this on my configuration:

voice translation-rule 3
rule 1 /23823867/ /4000/


voice translation-profile entrantes
translate called 3


dial-peer voice 106 voip
translation-profile incoming entrantes
session protocol sipv2
incoming called-number 23823867
codec g711ulaw
dtmf-relay rtp-nte
no vad

or this is a bad configuration?.

regards,

Juan Pablo

Problem connecting SIP Trunk to ISP

http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.

Problem connecting SIP Trunk to ISP

Hi,

i tried several configurations but without success.

this is the debug ccsip messages,,

this is when i dial... but when i test a call from outside to the CME i didn't get any log..

is there a way i could know the problem is the ISP sip server, or something like that??. so i could talk to the ISP and tell them that the problem is their configuration..

on the logs there is a line that says:

Received:

SIP/2.0 500 Server Internal Error

this is my CME reporting the error? or the ISP SIP server reporting the error?.

regards,

Jul 22 01:55:24.171: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:045118106@172.16.161.2:5060 SIP/2.0

Via: SIP/2.0/UDP 172.16.164.90:1114;branch=z9hG4bK-d8754z-82780a6e1618f04d-1---d8754z-;rport

Max-Forwards: 70

Contact: <4105>

To: <045118106>

From: "PRB"<4105>;tag=9f755413

Call-ID: MDA4ZmM4MTIwOTk4MzA2ODgyYTdmODY5NjY3ODE0YWI.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE

Content-Type: application/sdp

Supported: replaces

User-Agent: 3CXPhone 6.0.20943.0

Content-Length: 408

v=0

o=3cxVCE 255171150 400780425 IN IP4 172.16.164.90

s=3cxVCE Audio Call

c=IN IP4 172.16.164.90

t=0 0

m=audio 40030 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

m=video 40028 RTP/AVP 34

c=IN IP4 172.16.164.90

a=rtpmap:34 H263/90000

a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;

a=sendrecv

Jul 22 01:55:24.179: //18248/9B973937813F/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.16.164.90:1114;branch=z9hG4bK-d8754z-82780a6e1618f04d-1---d8754z-;rport

From: "PRB"<4105>;tag=9f755413

To: <045118106>

Date: Mon, 22 Jul 2013 01:55:24 GMT

Call-ID: MDA4ZmM4MTIwOTk4MzA2ODgyYTdmODY5NjY3ODE0YWI.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M2

Content-Length: 0

Jul 22 01:55:24.179: //18250/9B973937813F/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:045118106@172.23.7.134:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.3.84:5060;branch=z9hG4bK31F1663

From: "PRB" <23823867>;tag=3F00E2DC-1256

To: <045118106>

Date: Mon, 22 Jul 2013 01:55:24 GMT

Call-ID: 9B98721F-F1A811E2-81458DE9-8D33915@172.25.3.84

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2610379063-4054323682-2168425961-0148060437

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1374458124

Contact: <23823867>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 347

v=0

o=CiscoSystemsSIP-GW-UserAgent 7398 5918 IN IP4 172.25.3.84

s=SIP Call

c=IN IP4 172.25.3.84

t=0 0

m=audio 17834 RTP/AVP 0 19

c=IN IP4 172.25.3.84

a=rtpmap:0 PCMU/8000

a=rtpmap:19 CN/8000

a=ptime:20

m=video 17836 RTP/AVP 34

c=IN IP4 172.25.3.84

b=TIAS:85000

a=rtpmap:34 H263/90000

a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

Jul 22 01:55:24.183: //18250/9B973937813F/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.25.3.84:5060;branch=z9hG4bK31F1663;rport=64861

Call-ID: 9B98721F-F1A811E2-81458DE9-8D33915@172.25.3.84

From: "PRB"<23823867>;tag=3F00E2DC-1256

To: <045118106>

CSeq: 101 INVITE

Content-Length: 0

Jul 22 01:55:24.183: //18250/9B973937813F/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 500 Server Internal Error

Via: SIP/2.0/UDP 172.25.3.84:5060;branch=z9hG4bK31F1663;rport=64861

Call-ID: 9B98721F-F1A811E2-81458DE9-8D33915@172.25.3.84

From: "PRB"<23823867>;tag=3F00E2DC-1256

To: <045118106>;tag=b88652ba

CSeq: 101 INVITE

Content-Length: 0

Jul 22 01:55:24.187: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:045118106@172.23.7.134:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.3.84:5060;branch=z9hG4bK31F1663

From: "PRB" <23823867>;tag=3F00E2DC-1256

To: <045118106>;tag=b88652ba

Date: Mon, 22 Jul 2013 01:55:24 GMT

Call-ID: 9B98721F-F1A811E2-81458DE9-8D33915@172.25.3.84

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Jul 22 01:55:24.187: //18248/9B973937813F/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 172.16.164.90:1114;branch=z9hG4bK-d8754z-82780a6e1618f04d-1---d8754z-;rport

From: "PRB"<4105>;tag=9f755413

To: <045118106>;tag=3F00E2E4-F9B

Date: Mon, 22 Jul 2013 01:55:24 GMT

Call-ID: MDA4ZmM4MTIwOTk4MzA2ODgyYTdmODY5NjY3ODE0YWI.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M2

Reason: Q.850;cause=41

Content-Length: 0

Jul 22 01:55:24.199: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:045118106@172.16.1ex61.2:5060 SIP/2.0

Via: SIP/2.0/UDP 172.16.164.90:1114;branch=z9hG4bK-d8754z-82780a6e1618f04d-1---d8754z-;rport

Max-Forwards: 70

To: <045118106>;tag=3F00E2E4-F9B

From: "PRB"<4105>;tag=9f755413

Call-ID: MDA4ZmM4MTIwOTk4MzA2ODgyYTdmODY5NjY3ODE0YWI.

CSeq: 1 ACK

Content-Length: 0

Jul 22 01:55:35.363: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

VIP Super Bronze

Problem connecting SIP Trunk to ISP

You should contact your provider. They are disconnecting the call..

Received:

SIP/2.0 500 Server Internal Error

Via: SIP/2.0/UDP 172.25.3.84:5060;branch=z9hG4bK31F1663;rport=64861

Call-ID: 9B98721F-F1A811E2-81458DE9-8D33915@172.25.3.84

From: "PRB"<23823867>;tag=3F00E2DC-1256

To: <045118106>;tag=b88652ba

CSeq: 101 INVITE

Content-Length:

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