Hi,

Thanks for the immediate reply.

You are right ,using the dispatcher module , i am able to send the OPTIONS packet to MSC Telco.

But as i describer in my earlier mail, i am using the same dispatcher module to establish the sip trunk between my My Kamailio server and my Asterisk server.

There is a table in the database with the name dispatcher.

Now, in that table i have 2 records

one is my Telco SIP IP and the other is Asterisk PBX IP.

But as per my understanding from the google, dispatcher module is used for load balancing between the servers

Telco SIP server will be sending the calls to Kamailio and Kamailio has to distribute completely to Asterisk server instead of distributing the calls between Telco SIP IP and Asterisk.

Please help with it.

Warm Regards,
Sandeep Chakravarthi.

On Tue, Jul 14, 2015 at 10:28 PM, SamyGo <govoiper@gmail.com> wrote:

Hi,
You're right about using IP Auth in Kamailio. You'll need to use the permissions module. However I believe permissions module wont send the OPTIONS to the MSC SIP Server. For this you may alternatively use the "dispatcher" module.

Take a look at the sample kamailio.cfg here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

Follow the tag WITH_IPAUTH and I'm sure you'll be able to implement it easily.

BR,
Sammy

On Tue, Jul 14, 2015 at 12:51 PM, Sandeep Chakravarthi <ivschakravarthi@gmail.com> wrote:

Hi,
We have a requirement with one of our telco
We are using asterisk in our servers and we are planning to implement SIP-I protocol and we choosed kamailio for it.

In Kamailio website, i came to know that kamailio will be supporting both SIP-I and SIP-T protocols

Below is what we need and pls confirm whether it is possible or not?

Asterisk PBX <-------> Kamailio <--------> Telco MSC

Telco will be forwarding the calls to kamailio on sip-i protocol and kamailio server has to forward the calls to our Asterisk server by converting sip-i to standard sip protocol

Similiarly Asterisk will be initiating sip call to kamailio server and kamailio server should convert it into SIP-I and should forward the call to Telco MSC

1. I am able to establish the SIP trunk [sending OPTIONS from asterisk and kamailio acknowledges with 200 OK] between Asterisk and Kamailio using dispatcher module in kamailio and sip.conf in asterisk.

How to establish the SIP trunk between kamailio and telco MSC?
[Generally MSC will act as SIP server and kamalio should send OPTIONS packet and MSC will acknowledges with 200 OK]

My telco MSC has only provided me the MSC SIP IP and there were no username/passwords provided.
Means i need to use IP based authentication for the SIP Trunk establishment.

In Kamailio how to achieve it?

Please help and any suggestions/feedback will be highly appreciated and thankful

Regards,
Sandeep

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