Hi,
Can you please tell me under which route i need keep those values.
Whether in the RELAY route or ASTERISK rotue?
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 11:03 PM, SamyGo govoi...@gmail.com wrote:
1 - Take a look at the Kamailio transformations and
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi ,
Sorry for the delayed reply.
I have configured my
Hi,
Kamailio is sending 404 Response and its not MSC.
If you see the pcap file , Kamailio has to forward the SIP invite packet to
MSC which it got from Asterisk server. But it is not happening.
I am attaching the pcap one more time for your reference.
In my pcap, below are the server details
Thats because your configuration file is not sending packet out (RELAY) to
MSC instead it is only doing a Loadbalancer / destination lookup in
TOASTERISK route and comes out of it, processes the following routes in
order
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
route(PSTN);
1 - Take a look at the Kamailio transformations and psuedo-variable page.
change the $td to the IP of the MSC; modify the $ru as $rU + @
172.22.12.100:5060 where this is IP of MSC side.
2 - Wireshark guys could've said it SIP-3 - point is it doesnt matter at
this point since you know your MSC
Yes, You are right and done the changes as you suggested.
Kamailio server is forwarding the call to MSC. But two issues are there.
1 .In the INVITE packet which is being sent from kamailio server to MSC, it
is coming Request-Line: INVITE sip:0730092190@*172.22.14.12*
That is my kamailio server
Hi ,
Sorry for the delayed reply.
I have configured my Asterisk and kamailio server, but when i initiate one
outbound call from my asterisk server to kamailio server, kamailio server
is initiating the call to MSC.
Please find the attached pcap details for your reference.
Below is my kamailio debug
Below is output from the dispatcher table, Set-2 is a pool of asterisk
servers to be Load balanced, and Set-1 is the Telco IP.
KAMSBC01:~# kamctl dispatcher dump
SET_NO:: 2
*SET:: 2 *
URI:: sip:192.168.0.150:5050 flags=AP priority=1 attrs=
URI:: sip:192.168.0.151:5060 flags=AP
Hi,
Can you share the sample code to differentiate the both telco IP and our
server IP?
.
Warm Regards,
Sandeep Chakravarthi.
On Tue, Jul 14, 2015 at 10:55 PM, SamyGo govoi...@gmail.com wrote:
Sure but if you look into the dispatcher module there is a field called
'setid' or groupid. Use
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.
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
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:
Sure but if you look into the dispatcher module there is a field called
'setid' or groupid. Use it wisely to differentiate between the Load
Balanced asterisk pool and the Telco IP.
The dispatcher module is exactly what you should use. You can find out if
incoming source IP belongs to a particular
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