Hi there,
I am pretty new to all this (kamailio..)
I am trying to set up the following:
1. Kamailio with web-gui -> done
2. Asterisk / elastix for connecting to sip-provider / PSTN-> done
This works perfectly incoming / outgoing
3. Connecting elastix/asterix to kamailio -> done
I already have extensions configured on kamailio. They can call each other perfectly.
Now I do need a route-config or dialplan that routes ALL outgoing calls for all extensions to the
Extension of the asterisk/elastix-machine.
Is that possible and if so, how ?
I do need to provision more than a few hundred sip-subscribers in the near future so connecting
Them using just asterisk is not an option..
Thanks in advance, I really appreciate your help
Regards
lutz
On Tuesday 15 June 2010, maybelater wrote:
I am trying to set up the following: [..] 3. Connecting elastix/asterix to kamailio -> done
I already have extensions configured on kamailio. They can call each other perfectly. Now I do need a route-config or dialplan that routes ALL outgoing calls for all extensions to the Extension of the asterisk/elastix-machine.
Is that possible and if so, how ?
I do need to provision more than a few hundred sip-subscribers in the near future so connecting Them using just asterisk is not an option….
Hi Lutz,
sure this is possible. If you route all calls to one target, just rewrite the request of passing messages to the appropriate target (in this case the asterisk). In order to decide if the customer is allowed for a outbound routing you can authentificate them with the auth modules, or just look in the database, they are also several modules to help you here. If you want to have more flexibility for routing you could use a module like the dispatcher, lcr or carrierroute.
Cheers,
Henning
Hi Henning, thanks for the follow-up. Seems like I am way too behind on all this. "rewriting messages to the appropriate target...".. so how exactly does that work ?
Geeze I wish I was a coder... Thanks lutz
-----Ursprüngliche Nachricht----- Von: Henning Westerholt [mailto:henning.westerholt@1und1.de] Gesendet: Donnerstag, 17. Juni 2010 17:41 An: sr-users@lists.sip-router.org Cc: maybelater Betreff: Re: [SR-Users] kamailio dialplan
On Tuesday 15 June 2010, maybelater wrote:
I am trying to set up the following: [..] 3. Connecting elastix/asterix to kamailio -> done
I already have extensions configured on kamailio. They can call each other perfectly. Now I do need a route-config or dialplan that routes ALL outgoing calls for all extensions to the Extension of the asterisk/elastix-machine.
Is that possible and if so, how ?
I do need to provision more than a few hundred sip-subscribers in the near future so connecting Them using just asterisk is not an option….
Hi Lutz,
sure this is possible. If you route all calls to one target, just rewrite the request of passing messages to the appropriate target (in this case the asterisk). In order to decide if the customer is allowed for a outbound routing you can authentificate them with the auth modules, or just look in the database, they are also several modules to help you here. If you want to have more flexibility for routing you could use a module like the dispatcher, lcr or carrierroute.
Cheers,
Henning
On Thursday 17 June 2010, maybelater wrote:
Hi Henning, thanks for the follow-up. Seems like I am way too behind on all this. "rewriting messages to the appropriate target...".. so how exactly does that work ?
Hi maybelater,
normally one just rewrite the request URI of the INVITE msg that you are processing, this can be done by simple assignment in the cfg, e.g. with $ru, or other modules like the three i quoted earlier. Then you just forward() it stateless or t_relay() it stateful to the destination.
Cheers,
Henning
-
Henning Westerholt -
maybelater