[SR-Users] Proxy topology question

[Greg Mannie]

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.mihaylov at 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 <greg at 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.mihaylov at 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.**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 <greg at 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
>>>>
>>>>
>>>>
>>>>
>>>>
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>>>>
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>>>>
>>>>
>>>
>>
>>
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