Thanks for clearing the doubts.
You are very right, i am using kamailio as Media Relay.
Can you send me some specific document URL, from where i can
configure Asterisk as PSTN Gateway.
Can we set Kamailio and Asterisk in one server.
Thanks in advance.
Vijay
Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote:
On 22.08.2012 14:26, Vijay Thakur wrote:
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server.
Kamailio is a SIP proxy, not a media server. Maybe you mean that
you are using Kamailio with rtpproxy as media relay.
All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for
Outbound
Calls. For this purpose i have followed the web page :
If you wan to you Asterisk as PSTN gateway only, then there is no
need to follow this tutorial. This tutorial makes strong
integration of Kamailio and Asterisk. For PSTN gateway
functionality there is no need to integrate Kamailio and Asterisk
- just configure Asterisk as gateway and forwards PSTN calls from
Kamailio to Asterisk (and vice versa)
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
In this page, some points are not clear for me , as given below:
(1) In case you use *sipregs* you have to create a record for
each
extension where to set the 'name' to value of 'name' from
*sipusers*.
The rest is populated by Asterisk from registrations.
>
(2) Be sure you configure Asterisk *to not
authenticate* SIP requests
coming from Kamailio.
I am not sure that my local users chat is working through
kamailio or
asterisk, who is used for authorization.
What do you mean with "not sure"? For instant messaging between
users there is no need to use Asterisk.
In above setup the authentication is done by Kamailio only.
regards
Klaus
Any specific Web page to correct the issue
will highly appreciated
according to my scenario.
Kindly guide me. Thanks in advance.
--
Best Regards,
Vijay Thakur
(Assistant Manager - Networks)
Mobile : +91 8744018065
Mail :vijay.thakur@loopmethods.com
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