[RESOLVED] SIP Overspill route through PSTN

Asterisk Asterisk Support
1

Hi there,

I was looking for some guidance again if possible… I have a SIP trunk which allows me 15 concurrent calls and I also have 15 lines through PSTN. Ultimately, I want all the outbound calls to go through the SIP trunk, however if I have a maximum 15 concurrent calls is there a way I can route a 16th call through the PSTN?

I understand I could count the calls as they are made from my crm application and when it reaches 15 prefix the number called with a 9 which then routes the outbound through the pstn. But is there a way to handle this via the dial plan?

Thanks
Chad

Vicibox 5 64bit from .iso | Vicidial 2.8-416a | Build 131016-2112 | Asterisk 1.8.23.0-vici | Single Server

2

Managed to find how to do this using the GROUPS functions.

Thanks