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New Member

SRST and CUCM Business Edition

Hello All. I am trying to figure out how SRST mode works in conjunction with my CUCM BE 5000 server. I am currently using MGCP from call manager to my gateway, which is also my SRST router. To my understanding when the call manager box goes down the router switches from MGCP to H.323? Also, what happens to active calls? Do they get disconnected or do they stay connected? If I test this setup does the gateway itself need to lose connectivity with call manager or will an individual phone connect to the SRST gateway? I am not certain that the gateway can still be up with call manager using MGCP and an individual phone use the same gateway for SRST. I though about breaking connectivity between a test phone and call manager but not between the test phone and the gateway.

Thanks!

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1 ACCEPTED SOLUTION

Accepted Solutions
Hall of Fame Super Silver

SRST and CUCM Business Edition

Correct, you will need to configure the GW with call-manager-fallback, and dial-peers.

As far as the calls staying up it depends on the circuit type you have in place, with PRI circuit the call will stay up.

To test it since the GW is MGCP you will need to ensure GW can no longer communicate with CUCM as well, that is when it will transition from MGCP D-channel being backhauled to CUCM into SRST H323.

HTH,

Chris

8 REPLIES
Hall of Fame Super Silver

SRST and CUCM Business Edition

Correct, you will need to configure the GW with call-manager-fallback, and dial-peers.

As far as the calls staying up it depends on the circuit type you have in place, with PRI circuit the call will stay up.

To test it since the GW is MGCP you will need to ensure GW can no longer communicate with CUCM as well, that is when it will transition from MGCP D-channel being backhauled to CUCM into SRST H323.

HTH,

Chris

New Member

SRST and CUCM Business Edition

Thanks Chris! Also, I have two 2911 routers that I am using for SRST and two device pools that I will assign to the appropriate phones. On each 2911 I have a PRI circuit that will be used during SRST. My question is as follows:

When in SRST mode can I route outbound calls from the first gateway to the second gateway? For example phones that register to the first 2911 I would like to route outbound calls to the second 2911. I assume I would set up a dial peer on the first 2911 that routes calls to the second gateway and from there setup a dial peer that routes out the PRI. Also, can you setup priority and have outbound calls try the 2nd gateway for outbound calls first and then try the first gateway if for some reason the second one is not available.

Thanks,

Phil

Hall of Fame Super Silver

SRST and CUCM Business Edition

Yes, you can build voip dial peers with lower preference that point to the other GW.

HTH, please rate all useful posts!

Chris

New Member

SRST and CUCM Business Edition

Great! I assume the config would be similiar to the following:

On first 2911

dial-peer voice 100 voip

preference 1

destination pattern 9.......

session target ipv4:x.x.x.x

The second choice would be the local pri itself so can I assign preference to a Pots?

dial-peer voice 101 Pots

preference 2

destination-pattern 9.......

port 0/1/0:23

Hall of Fame Super Silver

SRST and CUCM Business Edition

Yes, if you skip preference it will be "0" which is highest priority then you can use "1" for the second one. You will also need the following:

voice service voip

allow-connections h323 to h323

HTH,

Chris

New Member

SRST and CUCM Business Edition

I do see voice service voip under the main config but it is not under the call-manager-fallback. Do I need to add this under this section?

Hall of Fame Super Silver

SRST and CUCM Business Edition

It goes under the main config not srst.

Chris

New Member

SRST and CUCM Business Edition

Thanks for the info Chris! I finally had  a chance to test this and confirmed that it was setup properly. One feature that I did find during my tests was that starting with IOS 15.1(2)T cisco introduced toll-fraud prevention. My voip dial peers kept failing and it took my awhile to figure this one out! I added the necessary ip addresses to the trusted list under voice service voip.

Thanks,

Phil

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