On outbound, you can simply make a Route List, Route Group. Place the SIP trunk in first, then whatever your backup method is.
Similar to PRI first, then FXO for backup in the Route List.
We're tying the SIP trunks inbound to server-based equipment
. When those SIP trunks go down, we want to reroute inbound calls to a hunt g
roup for Cisco handsets.
Since its inbound, you will have to control that beyond callmanager. For example, if the trunk came in from Verizon (one trunk) and that trunk went down, you have no control of inbound. If you had (2) trunks from Verizon and the first failed, then the inbound calls (if configured by Verizon) would roll to the second trunk. You can run internal tests with CME and SIP trunks by creating Dial Peers. Higher the dial peer number, the more priority it gets.
Not sure if this helps at all.....
Thanks for your answers, but here's one last piece of information that may better clarify my question:
I'm talking about using internal SIP trunks between CCM and an internal third-party IVR we have. So, the Verizon trunks stay up, but the SIP trunk from CCM to the IVR (which normally dishes out the calls to the extensions) goes down. Can CCM be configured to recognize that none of the extensions picked up, and route the calls to a different hunt group?
The failover scenario here is that if the IVR goes down, but the extensions are just fine, CCM can recognize that the calls went unanswered and route the calls to agents using a hunt group that doesn't go through the IVR.
You could possibly send it back to CUCM in the route list as a second option, but change the outgoing number to the hunt group pilot then.
So it would look like this:
SIP Trunk to IVR Route pattern: 6500
Route List from above.
On the CUCM SIP route List, there is an option to change the outgoing number (transform or translate)
So whent the call goes out, it will be translated to whatever number you want to that end point. You can translate to say 7500.
CUCM SIP receives 7500
7500 is your Pilot Point to the hunt group.
Phones are in the Hunt Group.
I use this method for H323 gateways that go down, I reroute to PSTN. Typically the site code is 7digits, but at the Route list for PSTN, I change the digits to dial the PSTN number.
Thank you for the replies. I am working with Ken (the op) to get this up and running. I do have some questions, but first I will provide some additional details.
So here is the current test setup.
The only SIP interface is this scenario is to the IVR. I have 10 SIP Trunks built as part of the Route Group (SIP RG) to connect to an internal IVR. Using a Route List (SIP RL) with the only Route List Member as SIP RG. To route calls to the IVR we have the Route Pattern 88898 set to use SIP RL as the Gateway/Route List entry.
Calls to the IVR work like a champ, but if the IVR goes down I need to re-route the calls to an internal Cisco Hunt Group that has IP phones as members of the group.
Route Pattern: 88898 (SIP Trunk Dialing)
So here is what I did or my situation.
I have an h323 gateway heading out to all my remote sites. (hub and spoke) Honestly, it should be a Gatekeeper, but different story.
In my Route List, I have (2) Route Groups.
Route Group A is the H323 gateway
Route Group B is the local PRI.
So if someone dials 4005000 for this route pattner for the Route list, it will hit this list.
If they get a busy signal on the h323 gateway, the route list says go to second route group.
On this route Route Group for the PRI, I configred the called party transform mask to 915551212XXXX
This then sends the call out the local gateway route pattern.
Also check this service parameter. Im not sure if SIP can do this, but worth a shot. Maybe the reorder is not coming back correctly and it cant hit the second list.
Also, instead of rolling to another SIP trunk, for testing, trying send it out the PRi to your cell phone. At least you know the Route List/Group is working.
I really appreciate that last post. You put me on the right track and I now have it re-routing to a Hunt Group.
I do have a couple of Cisco specific SIP Trunk questions that you may have the answers to.
How many simultaneous calls can one SIP Trunk support? The reason for the question is there are other PBX's that require one SIP trunk for every call.
Is there a licensing requirement/limitation for the number SIP Trunk interfaces that can be provisioned on the CCM?
Thank you again for the help.