SRST questions

wilson_1234_2 · ‎12-26-2007

We have Cisco Call Manager implemented and our five remote branches are all using Cisco IP telephones.

Each remote router is configured with SRST. Each router has it's own PRI with a block of DID numbers.

The calls from branch to branch all go through the Main site CAll Manager via our MPLS data cloud and everything else

(Branch inbound/outbound calls outside our network) goes through the PRIs for the branches.

When we loose MPLS connection to the branches and they loos connectivity to Call Manager, they loose all call capability outside of their branch.

I can see in the branch router config that the phones are set up to get a DHCP address from the router and other parts of the voice config.

My understanding is that SRST is supposed to allow the router to act as a limited Call Manager to route calls when we loose data connectivity, but the branches cannot receive or place calls other than internal to the branch from IP Phone to IP Phone.

Is SRST capable of allowing calls using the PRI in these situations?

During Data network outages, the branch managers have to use Cell Phones to place calls.

It seems that if the PRI works and is part of the router config, it is a short jump from only internal calls working during data network outages to being able to place calls via the PRI.

Brandon Buffin · ‎12-26-2007

Yes, SRST is capable of routing inbound/outbound calls using a PRI during a WAN outage. Please post your config.

Brandon

wilson_1234_2 · ‎12-26-2007

Here is a typical branch config.

Brandon Buffin · ‎12-26-2007

You only need one dial peer with the following commands:

incoming called-number .

direct-inward-dial

I prefer to make this a seperate dial peer for clarity, but you can include the commands under an existing dial peer. Also, the dialplan-pattern command should be similar to:

dialplan-pattern 1 9616667... extension-length 4

This is assuming your extensions are 7000 - 7999 and match your DIDs.

After you make these changes, enable "debug voip dialpeer" and make incoming and outgoing calls. This will allow you to see if the calls are hitting the dial peers that you expect. Post the results if you would like.

Brandon

wilson_1234_2 · ‎12-26-2007

Thanks for the information.

So you are saying there are problems with the config?

And that this is the reason for SRST not working when the data link is down?

What about outbound calls from the branch ip phones during data link failures?

If so, why would this not have been completed during the install of the site (the voice config was contracted to a consultant)?

Does that make any sense to you not to have this part set up?

Maybe that is why, the extensions do not match the DID range of numbers, where did you get this from the config"

"This is assuming your extensions are 7000 - 7999 and match your DIDs."

Brandon Buffin · ‎12-26-2007

At the very least, there is redundancy and ambiguity in the config. I have not tried using the commands I previously mentioned under multiple dial peers, but I can see where it could cause a problem.

Outbound calls from the branch phones should function properly when in SRST mode.

SRST should be setup and working properly at branch locations. That being said, I'm not sure what your scope of work included with the consultant.

I was making some assumptions regarding your DIDs/Extensions based on the config. What are the correct values for these?

Brandon

wilson_1234_2 · ‎12-26-2007

The extensions are 2001-2015, the DIDs are all over the place, not a concurrent block of numbers.

My understanding is the SRST is supposed to be working fully when the data link goes down.

I was not here when this was put in place and tested (if it was).

It does not surprise me, this consultant has done poor work in the past and present. There are a lot of unfinished and partially working projects.

What tells you that SRST should be working for out bound calls?

Also, what about inbound calls when in SRST mode?

Brandon Buffin · ‎12-26-2007

With your extensions and DIDs not matching, I would recommend removing the dialplan-pattern command.

I didn't mean to indicate that outbound calls should be working with your current config - only that outbound calls should function in SRST with the proper configuration. I just tried to look at your config again, but cannot. Can you try to repost it?

For inbound calls, I would first remove the duplicated commands from your dial peers. Again, I like to use a seperate dial peer for the incoming dial peer, but this can be combined with another dial peer. It just doesn't need to be under multiple dial peers.

Brandon

wilson_1234_2 · ‎12-26-2007

Here it is, I removed it just to be on the safe side.

Thanks for all of the great information.

Rajesh Revuru · ‎12-26-2007

Wilson:

With your current configuration, your phones will register and you will have basic dial-tone but i am not really 100% sure that your outbound and inbound calls would work in SRST.

You would have to re-do the dial-peers

Here is the link for SRST Admin guide

http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_book09186a0080861003.html

wilson_1234_2 · ‎12-26-2007

Thanks for your input, I appreciate it.

So, the existing dial peers would not allow for outbound calls to hit the phones.

What tells you that the dial peers will have to be changed?

Does a dial peer have to be created for each DID and associated with a particulatr phone?

Can I set that up as an addition to the current config?

Rajesh Revuru · ‎12-26-2007

Wilson:

Check to see if if you have the following ( i cannot see your config file)

1. SRST configuration (type "Sh ccm-manager fallback-mgcp) It will tell you if SRST is enabled

2. Dial peeers -outbound (sh run | b dial-peer)

3. Dial peers -inbound

If you have the above, you have to test simulating MPLS connection loss.

HTH

Rajesh Revuru · ‎12-26-2007

Hello:

Yes when gateway is in SRST mode, it will allow the calls coming on PRI and route it to the end-stations. For example if carrier is sending last 4 of XXX-XXX-XXXX digits on PRI and if you have your phone extension as XXXX you should be good and doesnt need any changes for your gateway.

When brances looses MPLS connection, you can still do IP-to-IP phone calling but this will go over the PRI, and you have to setup the translation pattern on the gateway where you would translate internal extension to PSTN number and thereby going over the PRI.

HTH