[SR-Users] Kamailio with asterisk for outbound calls

[Klaus Darilion]

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 at loopmethods.com
>>>
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>