[SR-Users] Using Kamailio with a SIP Trunk

[Rainer Piper]

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 
> <mailto: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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