[SR-Users] Proxy topology question

[Greg Mannie]

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")||($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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>