On 23.08.2012 08:31, Vijay Thakur wrote:

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.

There is no such document. But configuring a PSTN gateway is already in the default configuration file. Just search in the deafult configuration file for "WITH_PSTN".

Can we set Kamailio and Asterisk in one server.

Yes, thats no problem. Either use 2 IP addresses on the same server, one for Kamailio and one for Asterisk, or use the same IP address and different ports.

regards Klaus

PS: If you are building a public SIP service it is a good idea to not use the default port 5060 to get rid of SIP port scanners.

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

Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web:www.loopmethods.com

LOOP Disclaimer

This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited.

---

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users