[SR-Users] Kamailio Infront of Asterisk with remote PBX

[Ovidiu Sas]

You can implement whatever routing logic you want in kamailio.
Also, you can use different sets of modules to implement same type of
routing logic.

The To header is irrelevant in SIP, no need for re-write.
When a call is received from an endpoint, based on it's IP address you
can choose what to do with the call (for example: find the right user
for the given extension).

You need to figure out all the requirements for your new setup and
then start implementing it.

Since you already have kamailio servers in your setup, you are
familiar with how kamailio works.
All you need to do is do some more modules README reading and figure
out which one is a better fit for your needs.


Regards,
Ovidiu Sas

On Thu, Oct 23, 2014 at 1:16 PM, Kenny Watson <KWatson at geniusppt.com> wrote:
> Hi Fred,
>
> Its more that the "user" on Kamailio is actually a PBX with extensions on it.
>
> On asterisk I'd usually do Dial(SIP/peername/extension) but I obviously cant do this as Kamailio is the peer that the call is being routed to initially.
>
> What I need to figure out is how to on kamailo maybe using a dial prefix specify that the call is going to a remote extension on a "user" and rewrite the to header to be extension at useripaddress rather than user at useripaddress.
>
> Does this make sense?
>
>
> Thanks
> Kenny Watson
>
> -----Original Message-----
> From: sr-users-bounces at lists.sip-router.org [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fred Posner
> Sent: 23 October 2014 16:34
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
>
> If you want to call a user on Kamailio from Asterisk...
>
> example...
>
> exten => s,1,Verbose(4,calling user on kamailio)
>   same => n,Dial(SIP/USERNAME at KAMAILIO,time,options)
>   same => n,--after dial logic --
>
>
> Fred Posner
> The Palner Group, Inc.
> http://www.palner.com (web)
> +1-503-914-0999 (direct)
> +1-954-472-2896 (fax)
>
> On 10/23/2014 11:21 AM, Kenny Watson wrote:
>> Hi Fred,
>>
>> Thanks for the quick response.  I already do use some Kamailio
>> features on our internal network for load balancing.
>>
>> The use case that I'm interested in is to effectively replace an
>> asterisk server that I use for SIP trunking to remote phone systems
>> with a Kamailio registrar/proxy and a bank of asterisk servers placing
>> calls direct to extensions on the remote PBX.
>>
>> I currently have this running on asterisk which I route to the
>> different remote PBX extensions using prefix based routing down to the
>> destination peer on asterisk which is essentially what I need to
>> replicate on Kamailio.
>>
>> i.e.
>>
>> 2021XXXX routes to   XXXX at remotepbx1
>>
>> remotepbx1 maybe defined as either by IP address or via a "normal"
>> registered sip peer with a username/password combo.
>>
>>
>> I understand that I can dial a registered device directly but its how
>> to call a remote extension on a registered device via Kamailio.
>>
>> Thanks Kenny Watson
>>
>>
>>
>>
>>
>> -----Original Message----- From:
>> sr-users-bounces at lists.sip-router.org
>> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Fred
>> Posner Sent: 23 October 2014 16:00 To: sr-users at lists.sip-router.org
>> Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
>>
>> Hi Kenny,
>>
>> This depends on the carriers and scenarios that you may use. I know
>> "depends" is a horrible answer, but one of the great aspects of
>> Kamailio is the flexibility of the modules.
>>
>> Some deployments may have a group of Asterisk servers all configured
>> similarly for handling calls. With this type of scenario, you would
>> benefit from using the dispatcher module.
>>
>> Many people like to use Kamailio on the public side of their network
>> and keep their asterisk servers on the private. This would be an
>> example of when to use rtpproxy (in bridge mode).
>>
>> Some carriers hate seeing the chain of systems on your network (ie the
>> asterisk boxes). Sometimes the use of TOPOH helps to integrate with
>> the carriers who have chosen their own "interpretations" of RFC for
>> "security."
>>
>> And there's more...
>>
>> The bottom line, is that the devil is in the details.
>>
>> Fred Posner The Palner Group, Inc. http://www.palner.com (web)
>> +1-503-914-0999 (direct) +1-954-472-2896 (fax)
>>
>> On 10/23/2014 09:12 AM, Kenny Watson wrote:
>>> Hi,
>>>
>>>
>>> I have a few asterisk servers providing some basic SIP trunking and
>>> routing.
>>>
>>> We have remote PBXs trunked onto asterisk which calls come into
>>> asterisk and are routing down to extensions on the remote PBX via
>>> prefix routing.
>>>
>>> I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer
>>> which Invites come into and are routed out to either SIP phones which
>>> are registered or to the remote PBX.
>>>
>>> I'm looking for some advice as to which modules would be best to use
>>> to achieve this as the remote PBXs will be dynamically registered
>>> rather than fixed gateways.
>>>
>>> Please let me know what further information would be helpful.
>>>
>>> Thanks
>>>
>>> Kenny Watson
>>>
>
> _______________________________________________
> 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



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com