SIP Trunk ITSP

[clark white]

Hello,

I am trying to setup a new sip trunk (new connection) to ITSP, whenever i dial a PSTN number i get a fast busy tone, My ITSP says everything is OK from their end,

Can anybody refer me for step by step troubleshooting docs for such type of issue.

[Ayodeji Okanlawon]

Clark,

You start by looking at the sip traces..

debug ccsip messages

You can send here and we can look at it.include calling and called number

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

Hello

here is the attached

calling number 8243

called number 244892969

Thanks

[Ayodeji Okanlawon]

What or who is this IP ? 10.230.0.26

This device is sending internal error..

Received:

SIP/2.0 500 Server Internal Error

Via: SIP/2.0/UDP 10.240.23.235:5060;branch=z9hG4bK8B80;rport=59611

Call-ID: 8536818-AB3211E2-A190F940-B2641F7D@10.240.23.235

From: <8423>;tag=24C9527C-10F8

To: <24489269>;tag=chkxe7ov

CSeq: 101 INVITE

Content-Length: 0

Can you send your sh run please and explain the ips in it

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

Hello,

Attached is the sh run

1 question is it compulsory to create a SIP trunk between VG and callmanager when a SIP trunk is between VG and ITSP.

PRI SIP Server ITSP: 10.X.X.X

CallManager: 10.X.X.X

VG: 10.X.X.X

[Ayodeji Okanlawon]

It is better to use sip to sip  set up. So yes create a sip trunk betwwen cucm and vg..and then point your route pattern to the sip trunk..

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

But is this a issue or issue is lying somewhere else

any configuration example to create a sip trunk between a VG and CUCM

[Ayodeji Okanlawon]

It is easy..

Go to CUCM, device>sip trunk>add new

Add all the required parameters, device pool, inbound CSS and then specify the destination ip as the ip address of the voice gateway..The local interface...

Then configure your route p atterns to the sip trunk

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

Here is the attached after i changed to SIP trunk

[Ayodeji Okanlawon]

At the moment, you have configured your inbound dial-peer from cucm to VG to use sip protocol. However you are using h323 from your CUCM to your gateway..this wont work...So you either configure a sip trunk which is better or reconfigure your gateway to use h323 on the inbound leg

dial-peer voice 10 voip

preference 1

destination-pattern .T

session protocol sipv2

session target ipv4:10.230.16.26

incoming called-number .

dtmf-relay rtp-nte

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

please find the attached debug,

the results are same

[Ayodeji Okanlawon]

configure this..

voice service voip

sip-ua

early-offer forced

Test again and send debugs

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

there is no such command under

sip-ua

early-offer forced

It is in dial-peer should i remove from there???

also i have changed

dial-peer voice 1 voip

destination-pattern 8...

session protocol sipv2

session target ipv4:10.X.X.X

incoming called-number .

no voice-class sip early-offer forced

no voice-class sip pass-thru headers

no voice-class sip pass-thru content unsupp

dtmf-relay rtp-nte

[Ayodeji Okanlawon]

sorry its..

voice service voip

sip

early-offer forced

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[clark white]

The result are same.please find the attached