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Multiple Registrar Servers/Open NAP/Domains

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Multiple Registrar Servers/Open NAP/Domains Empty Multiple Registrar Servers/Open NAP/Domains

Post by Guest Thu Apr 29, 2021 7:08 am


I have an Open NAP configured for remote extensions to register to a set of FreeSWITCH instances sitting behind ProSBC. I have a SIP domain configured in ProSBC called “” and I have an extension “102” registered successfully to Freeswitch via ProSBC on this domain.

When I call outbound from the phone registered on extension 102, ProSBC is not sending the call towards the Freeswitch server that the domain is configured to register to, but rather is round-robin based on routes configured in ProSBC. I have two routes for calls coming from the open NAP towards each of my two Freeswitch servers. I am attaching screenshots of the routes.

I would expect my handsets to send all calls to the Freeswitch instance it is registered to, but this is not happening. Additionally, if I call from a registered handset to a phone number that there is a route for in ProSBC, it is never touching Freeswitch and ProSBC is directly attempting to route it.

Are there any settings that can be changed in ProSBC to aid in this?

Multiple Registrar Servers/Open NAP/Domains Open_f10

Multiple Registrar Servers/Open NAP/Domains Open_f11


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Multiple Registrar Servers/Open NAP/Domains Empty Re: Multiple Registrar Servers/Open NAP/Domains

Post by Guest Thu Apr 29, 2021 7:11 am


You should follow the below link for domain creation, registration, and how to set the route from OPEN_NAP to Freeswitch for registration as well as CALL INVITE, as well as how to route call back to the registered user (the phone) on the internet.

Here, you have open_NAP two Freeswitch routes, to freeswitch1 and to freeswitch2, and there is no condition to distinguish between these two routes, so the call will match on two routes, and round-robin as you see to catch the first route and if not successful, the second route.

In order for the call to terminate exactly on freeswitch1 or freeswitch2, you may need to match it separately using something like calling matching, as the regular expression in the calling field of the route, for example,


Calling means From field of SIP INVITE.   So that when eg. calling is with, the call will be terminated on freeswitch1 only.

For SBC use case and configuration, you could also refer to below,



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