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<div class="moz-cite-prefix"><big>Thanks for clearing the doubts.
You are very right, i am using kamailio as Media Relay. <br>
Can you send me some specific document URL, from where i can
configure Asterisk as PSTN Gateway. <br>
Can we set Kamailio and Asterisk in one server. <br>
<br>
Thanks in advance. <br>
<br>
Vijay <br>
</big><br>
Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote:<br>
</div>
<blockquote cite="mid:5035C583.1090806@pernau.at" type="cite">
<br>
<br>
On 22.08.2012 14:26, Vijay Thakur wrote:
<br>
<blockquote type="cite">Hi All Kamailio Experts,
<br>
<br>
I have configured Kamailio (kamailio 3.1.5) as media server.
<br>
</blockquote>
<br>
Kamailio is a SIP proxy, not a media server. Maybe you mean that
you are using Kamailio with rtpproxy as media relay.
<br>
<br>
<blockquote type="cite">All things
<br>
are working fine. Now i want to use Asterisk (Asterisk 1.6) for
Outbound
<br>
Calls. For this purpose i have followed the web page :
<br>
</blockquote>
<br>
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)
<br>
<br>
<blockquote type="cite"><a class="moz-txt-link-freetext" href="http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb">http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb</a>.
<br>
In this page, some points are not clear for me , as given below:
<br>
<br>
(1) In case you use *sipregs* you have to create a record for
each
<br>
extension where to set the 'name' to value of 'name' from
*sipusers*.
<br>
The rest is populated by Asterisk from registrations.
<br>
</blockquote>
>
<br>
<blockquote type="cite">(2) Be sure you configure Asterisk *to not
authenticate* SIP requests
<br>
coming from Kamailio.
<br>
<br>
I am not sure that my local users chat is working through
kamailio or
<br>
asterisk, who is used for authorization.
<br>
</blockquote>
<br>
What do you mean with "not sure"? For instant messaging between
users there is no need to use Asterisk.
<br>
<br>
In above setup the authentication is done by Kamailio only.
<br>
<br>
regards
<br>
Klaus
<br>
<blockquote type="cite">Any specific Web page to correct the issue
will highly appreciated
<br>
according to my scenario.
<br>
<br>
Kindly guide me. Thanks in advance.
<br>
<br>
--
<br>
Best Regards,
<br>
<br>
Vijay Thakur
<br>
(Assistant Manager - Networks)
<br>
Mobile : +91 8744018065
<br>
Mail :vijay.thakur@loopmethods.com
<br>
<br>
Loop IT Methods Private Limited
<br>
1st Floor, B-10, Sector-7, Noida, (U.P) India
<br>
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390
011 178 (AUS)
<br>
Fax: +91 971 728 330
<br>
Web:www.loopmethods.com
<br>
<br>
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