I am using a 2821 one as a gateway to route PSTN PRI calls to/from my SIP enabled PBX. In the even that my PBX is not available, I need to route my PSTN calls to a different PBX. I am using preferenced dial-peers to do so. The problem I am experiencing is that the roll over to the next dial-peer is not fast enough. The PSTN will drop my call before the 3rd (Or even the 2nd) dial-peer can setup the SIP connection.
I need to either:
a. - have my dial-peers monitor their session target and place them self in a "down" state when the target is not reachable
b. - find a way to have the dial-peers time out to the next preference much faster than they currently do.
c. - Have the gateway preliminarily answer the call and provide some sort of ring back to the Telco while it runs through the dial-peers to find an operative choice. (this one is probably not gonna happen)
I have reasearched and found some solutions which almost work, such as call fall back. But from what I've read on call fall back, it seems to only work if you're monitoring another cisco device which is configured to respond to your probes. Which I'm not.
My gateways perform very basic call routing, with everything is is handled by the PBX. I don't have SRST, but I will get it if someone can tell me it will solve my problems. Here is an example of 3 dial-peers I have setup via preference.
dial-peer voice 1 voip description All 5555555 Calls to PBX A
I'm not sure if it is normal procedure to reply to your own post with a solution, but here goes...
First I setup some IP SLA monitoring on my primary and secondary phone servers:
ip sla 201 icmp-echo 10.x.x.1 source-interface Loopback1 ! ip sla 202 icmp-echo 10.x.x.2 source-interface Loopback1 ! ip sla group schedule 200 201-202 schedule-period 2 frequency 10 start-time now life forever
! The group schedule is not necessary. Having my monitoring synced up just satisfies my anal retentive tendencies
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