[SR-Users] Using Kamailio with a SIP Trunk

Salman Zafar msalman212 at gmail.com
Wed Mar 26 17:41:37 CET 2014


Hi,
   Well, you do not really need any trunk at Kamailio, all you have to do
is get your NGN to be able to allow access to your Kamailio IP address, and
it can SIP PING it i.e sending OPTIONS to see if it alive, if required.  It
will do (Kamailio to NGN calls), similarly you can do the same for NGN to
Kamailio calls as well, your Kamailio sever should be able to authenticate
your NGN IP and do the routing as per your requirement.


On Wed, Mar 26, 2014 at 9:32 PM, Rizwan Khan <rizkhan at gmail.com> wrote:

> Thanks for the help.
>
> Is it possible to have a direct sip trunk from kamailio to an NGN without
> involving asterisks? I will be using NGN to route outside calls for
> landlines and mobile.
> On Mar 24, 2014 9:37 PM, "Rainer Piper" <rainer.piper at soho-piper.de>
> wrote:
>
>>  Hi Rizwan,
>>
>> that is the right approach .
>>
>> For adding an Asterisk as SBC you can use the section route[PSTN] at
>> kamailio.cfg
>>
>> #!ifdef WITH_PSTN
>> # PSTN GW Routing
>> #
>> # - pstn.gw_ip: valid IP or hostname as string value, example:
>> # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
>> #
>> # - by default is empty to avoid misrouting
>> pstn.gw_ip = "IP_OF_YOUR_ASTERISK"
>> pstn.gw_port = "5060"
>> #!endif
>>
>> Make kamailio IP as trusted IP in your asterisk sip.conf like
>>
>> [kamailio]
>> type=friend
>> context=outgoing-kamailio
>> host=[IP_OF_YOUR_KAMAILIO]
>> port=5060
>> qualify=no
>> ;trustrpid=yes
>> ;sendrpid=yes
>> deny=0.0.0.0/0.0.0.0
>> permit=[IP_OF_YOUR_KAMAILIO]
>>
>> add outgoing trunk Data to your asterisk sip.conf and extensions.conf
>> section [outgoing-kamailio]
>>
>> and that is it.
>>
>>
>> Regards
>> Rainer
>>
>>
>>
>>
>> Am 24.03.2014 16:23, schrieb Rizwan Khan:
>>
>> Is my question not well phrased? Or is too general? Can anyone help with
>> a document or an older thread which could help me?
>>
>> Thanks
>> On Mar 24, 2014 1:57 PM, "Rizwan Khan" <rizkhan at gmail.com> wrote:
>>
>>>  I want the following setup:
>>>
>>>  1 Kamailio server to handle internal calls (A/V), IM and Presence.
>>> 1 Asterisk or any other way to communicate with an NGN where I will
>>> create the SIP Trunk to route calls outside of the network.
>>>
>>>  Is this the right approach or there is a way to directly communicate
>>> with the NGN to make a SIP trunk by using some external modules.
>>>
>>>  Any guidelines will be highly appreciated.
>>>
>>>
>>> Rizwan Khan
>>>
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> --
>> *Rainer Piper*
>> NOC - +49 (0)228 97167161 - sip.soho-piper.de
>> NOC - +49 (0)2247 9064188 - sip.tele33.de - sip.tefonix.de - D293
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Regards

M. Salman Zafar
VoIP Professional
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140326/3b778ac6/attachment.html>


More information about the sr-users mailing list