14 Jul
2015
14 Jul
'15
8:25 p.m.
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 set in dispatcher table thus you
can tell if call is coming from Telco or from your Asterisks.
You can select the dispatcher set for load balancing but if we only have
one IP in there then it gets all the load.
BR,
Sammy
On Tue, Jul 14, 2015 at 1:21 PM, Sandeep Chakravarthi <
ivschakravarthi(a)gmail.com> wrote:
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(a)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(a)gmail.com> wrote:
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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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