I suppose this is better for you:
http://kamailio.org/docs/modules/3.2.x/modules/carrierroute.html#id2548186
But I never used it - we just did not need anything alike.
>From overview - you can define routing tree per user.
On Mon, Feb 6, 2012 at 8:13 PM, Greg Mannie <g...@latigi.com> wrote:
> Thank you again for your help.
>
> I have been reading on the dispatcher module and have a question. It
> would seem many people use Kamailio as more than proxy and register sip
> extensions against it. Since we host virtual pbx (asterisk 1.8, Freepbx)
> for a few different clients, each instance is separate with it's own
> database.
>
> I am having problems wrapping my head around the configuration I should
> use. Is there not a method just to add DID in the same fashion as asterisk.
> So I register a trunk on Kamailio and based on incoming DID it sends it to
> the correct asterisk server?
>
> I have my asterisk 1.8 on a public ip address with a trunk registered to
> the Kamailio server which also has a public ip address.
>
>
> Regards,
>
> Greg
>
>
>
>
> Quoting Stoyan Mihaylov <stoyan.v.mihay...@gmail.com>:
>
> I dont know about siremis, but you can forward calls to different groups
>> of
>> Asterisk servers - using
>> ds_select_dst(set, alg);
>> Where set is set of Asterisk servers - you can check that module.
>> The problem is - I have no idea how you can select different sets in
>> kamailio.cfg, except by length, or some matching pattern in CallerID.
>> But if you put whole logic used in Asterisk in DB - then you dont care
>> which server will take the call, because you can put whole logic purely in
>> DB - including extensions etc.
>> At least I prefer to have almost nothing in extensions.conf - and
>> everything to stay either in DB or in AGI scripts.
>>
>> My knowledge of Kamailio is very very basic - I know only few things
>> there.
>> Asterisk and Kamailio can run on same server, but I cant see any reason
>> for
>> that. I mean you will have lot of troubles in such case, and nothing
>> "good". This is only if you want to make some tests. But you can expect
>> lot
>> of troubles.
>>
>>
>> On Fri, Feb 3, 2012 at 9:40 PM, Greg Mannie <g...@latigi.com> wrote:
>>
>> Thank you for your detailed response. Sorry for the trouble but would
>>> you
>>> be able to also answer the following.
>>>
>>> Do you know if this same type of deployment would be suited to our needs.
>>> Many of the Asterisk servers we host are for clients, who have their own
>>> extensions, voicemail, ivr etc. I was hoping I could setup routes on the
>>> kamailio and direct them to the appropriate asterisk server.
>>>
>>> Initially I thought it would be as simple as setting up an inbound route
>>> on Asterisk. Ha.. I also installed siremis 3.2 and perhaps reading on
>>> how
>>> to use it will provide clearer details.
>>>
>>> I know so little, I'm not even sure if I need to have Kamailio and
>>> Asterisk running on the same server, since I only want Kamailio as a
>>> proxy.
>>>
>>>
>>> Regards,
>>>
>>> Greg
>>>
>>>
>>> Quoting Stoyan Mihaylov <stoyan.v.mihay...@gmail.com>:
>>>
>>> We were in similar situation. Many years with Asterisk and then we were
>>>
>>>> forced to use ser - and we preferred Kamailio.
>>>> Now we do:
>>>> Kamailio has global IP address and clients register to it.
>>>> Kamailio forward all calls to Asterisk boxes using following:
>>>> ds_select_dst("1","4");#You can use many asterisk boxes this way
>>>> $sht(forw=>$ft)=$du; #this way I store used path
>>>> I used t_relay, instead of forward, because my Asterisks are with local
>>>> IP.
>>>> Calls from Asterisk are send to Kamailio if they are to local user, or
>>>> to
>>>> our SIP provider. There are no problems with calls from Asterisk to SIP
>>>> provider, even if Asterisk is behind NAT.
>>>> Asterisk accepts calls from SIP provider though registrar lines in
>>>> sip.conf. Asterisk can forward calls from our SIP provider to local
>>>> users
>>>> in Kamailio.
>>>> I got problems with ACK and BYE. To solve them, I used
>>>> if(($td=="sip.name.of.**kamail**io.server.com<http://kamailio.server.com>
>>>> <http://sip.name.**of.kamailio.server.com<http://sip.name.of.kamailio.server.com>
>>>> >
>>>> ")||($si=="**IPofServer")){
>>>>
>>>> $du=$sht(forw=>$ft);
>>>> }
>>>>
>>>> On Fri, Feb 3, 2012 at 8:13 PM, Greg Mannie <g...@latigi.com> wrote:
>>>>
>>>> Hello
>>>>
>>>>>
>>>>> After much reading I have come to the realization that after years of
>>>>> using Asterisk I know very little about Sip.
>>>>>
>>>>> I have my Kamailio box up, I have Asterisk 1.8.x running with realtime
>>>>> working. I thought it would be just a case of registering SIP trunks
>>>>> from
>>>>> my provider to the kamailio and registering our internal asterisk
>>>>> servers
>>>>> to the kamailio.
>>>>>
>>>>> Much of what I read talks about using Asterisk as the PSTN interface,
>>>>> but
>>>>> that interface is through a sip trunk purchased from a provider. Won't
>>>>> Kamailio be the PSTN gateway? The idea here is to pool all the sip
>>>>> trunks
>>>>> from the various hosted asterisk solutions (VM running asterisk) and
>>>>> point
>>>>> them all to a proxy to facilitate the aggregation of traffic.
>>>>>
>>>>> I have been reading SIP tutorials and the mailing list archives. If
>>>>> anyone has a sample config and perhaps a little direction it would be
>>>>> highly appreciated.
>>>>>
>>>>> Thank you
>>>>>
>>>>> Greg
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ______________________________******_________________
>>>>>
>>>>>
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>>>>> <http://lists.sip-**router.org/**cgi-bin/mailman/**listinfo/sr-**users<http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users>
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>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>
>>>
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>>
>
>
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