Group of Outgoing sip trunks

I have some fxo gws that are connected to asterisk server among sip trunks. I defined a group of this sip trunks and forward all outgoing calls to this group. How can I do something like Dahdi group for this sip trunk group?some thing like channel hunting, method of selection, state of trunk before dialing ?

thanks a lot.

Consult the people who are implement the SIP trunk concept. This is a concept imposed by things like GUIs. Asterisk has no such concept.

Generally you need to program them as explicit searches in the dialplan.

Thanks for reply.
So sip peer or sip trunk or any concept that is related to this. Asterisk has any function like DAHDI Groups for sip peers? if it doesn’t have anything for this, it will be a good solution to use dial plan in high capacity cases?

As david55 already told You: There are not meachanisms comparyble to DAHDI groups available for SIP-connections.
Nevertheless the functionality can be achieved by implementing the logic within Your dialplan. The “load balancing” or “availibility chaecking” for a SIP-connection is even easy. H

owever: In business-scenarios You would usually choose a channel-dimension with Your ITSP which makes it dispensable to do “load balancing” for just achieving a free line.
The standard use case for selecting a specific SIP-Line would normally be some kind of least coast routing or - in some very special cases - the reachability of certain national numbers when SIP-Lines in several countries are used (e.g. for the reachability of national TollFree or PremiumRate services). Other scenarions may occur in residential situation wher You’ve got only 2 concurrent calls on a line.

Thank you for your answer. I got it.
I have some sip trunks that they don’t have any priority or weight. but I want to dispatch the calls between all of them.
So is this possible with asterisk? Do you have any solution for asterisk?

I found a module (mod_distributor) in FS. Is there any app same this in Asterisk ?


FreeSwitch has a completely different architecture.

There is no need for this to be implemented in the core and the normal policy with Asterisk is not to implement things in the core if they can be programmed in the dialplan.

This situation normally only arises if you are trying to misuse single extension trunks for PABX use. A normal SIP endpoint can handle as many simultaneous calls as the underlying IP network can accommodate. There are no circuits, like there are with dahdi, that can only carry one call each.

Thanks for reply.

Yes exactly my situation is same. So when rate of calls is high, it is not better to be implemented in the core vs dialplan?
In dial plan, I should extract the status of trunks and after that, distribute a call between trunks as round robin,linear, based on weight or etc.

Can you provide a sample?!