SOLVED Redundant/Overflow SIP Trunks

[anomaly0617]

Hi there,

I have an organization where they have multiple sites with PIAF at each site, and SIP trunks from their Internet provider at each site. Each site has an internet circuit that goes to/from the hardware firewall and a SIP Session Border Controller that goes to a secondary interface on the PBX. Each site has IPSec VPN tunnels to all the other sites via the firewalls, so it's a "meshed" configuration.

At each location, the minimum number of SIP trunks the ISP would provide was either 6 or 8, depending on the ISP in question. However, at a few of the locations the chances of them actually USING all 6 or 8 channels in a given moment is pretty much non-existent.... and then at the other locations, there's the distinct possibility that during a busy time (say, the end of tax time) they could exceed the 8 SIP trunks for a few days out of the year. Not enough to add more SIP trunks, but also enough to annoy people when they get a busy signal trying to dial out.

I've already got IAX intra-company trunks established between the locations and outbound routing for those. If you dial a 5000 number from a location where the dial plan is in the 6000's, it will pick the outbound route with the IAX trunk and just connect the calls from PBX to PBX over the VPN tunnel. This way we don't burn up two SIP channels (one on each end) for a phone call, as long as the VPN tunnel and IAX trunk is alive.

Now what I'm interested in doing is potentially routing outbound call traffic from Site 1 through Site 2's SIP trunks, in an overflow fashion. So Outbound Route 1 would be to the local SIP trunk Session Border Controller. If it shows congestion, Outbound Route 2 would go to the next site's IAX trunk and attempt to locate a channel there. So, if a location with 8 SIP trunks tries to make a 9th outbound call, that call goes through one of the remote branch office's SIP trunk and the call still completes.

I did some testing on this today. When I set up a very specific outbound route (my extension, calling my cellular phone, routes to SIte 2) I get the receptionist at Site 2. I then tried an Inbound route at Site 2 stating that if my extension called in, route it to the SIP trunk. I still got the receptionist after trying.

Being that it's in the middle of the day and also in the middle of a busy season, this is where I gave up. But if someone has this running somewhere else, what's the secret to making it work?

This might be easier if all the locations had the same ISP provider. Unfortunately only two of the locations share the same provider. The others are all different, So it's not even like I can set up a big pool with the ISP and route all the calls through that pool. Similarly I've tried the independent SIP trunk carriers and gotten nothing but complaints from staff, managers, and the owners. They say calls drop out for 1-3 seconds in the middle of a 30 minute conversation and this just isn't acceptable. This was the catalyst for the shift to ISP-based SIP trunks. Our preferred independent SIP trunk provider could not offer me a workable long term solution (trust me I tried), so we shifted to ISP based SIP Trunks.

Thanks, in advance!
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Anomaly0617

 

[anomaly0617]

Solved - Eugene in this thread had the right answer. In the trunk setup for the IAX trunks, change the context=from-trunk to context=from-internal. This will allow the call to pass through. That being said, on mine they are still showing up as being from the site with the SIP trunk's Caller ID, even though I cleared the outbound caller ID information from the outbound route and trunk and put it directly in extensions. But this is an issue I can work out later.

So, thanks, Eugene!