[SR-Users] Using Kamailio with a SIP Trunk

[Rizwan Khan]

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
>>
>>
>>
>
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>
>
>
> --
> *Rainer Piper*
> NOC - +49 (0)228 97167161 - sip.soho-piper.de
> NOC - +49 (0)2247 9064188 - sip.tele33.de - sip.tefonix.de - D293
>
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