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Switching between two PRI lines if one gets down

Deep9769
Level 1
Level 1

Hi,

I'm quite new to the Cisco unified communications manager. In my premises I have two PRI connection at gateway 

Primary PRI => 022 4303 "extension number" for Incoming calls  and we need dial 90 "mobile number" for outgoing calls with reference to my Cisco up phone's

In similar way for 

Secondary PRI => we need dial from Cisco ip *90 "mobile number" for outgoing calls.

But I want to configure my gateway in such way so that it can automatically switch between PRI lines if any one PRI is down, so users doesn't need to change the dailing pattern if any one PRI is down or not working 

Requesting you to please suggest how I'll able to achieve this, I'm sure there are some parameters which I need configure in gateway but I don't know how to do it, please any one guide me

39 Replies 39

It seems that you have some confusion about terminology, a SIP trunk is not the same as a PRI. A PRI is an ISDN type of circuit that carries TDM traffic, it does not carry SIP traffic. I’ll get back to you on your dial peers when I’m in front of a computer.



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Reading what you wrote once more I see that you're using two different route patterns, 90 that sends calls to the gateway and uses the SIP trunk to you service provider and *90 that uses the MGCP controlled PRIs. To get an automatic switch between these two you would configure this.

  1. RP that matches whatever you want it to match, for example 90!
  2. A RG that contains the SIP trunk towards the gateway
  3. Another RG that contains the MGCP controlled PRIs on the gateway
  4. Put these RGs into a RL
  5. Configure whatever number modification you need on the RL/RG level

That's all there is to this, assuming that your gateway configuration is correct.

On the dial peers I would based on the configuration you shared recommend that you do these changes.

 

no dial-peer voice 102 voip !not needed as dial peer 101 is identical
no dial-peer voice 202 voip !not needed as dial peer 201 is identical
!
dial-peer voice 101 voip
 destination-pattern .T !Change this to something more specific than .T as that matches anything. I would put it as 90T if your using that as the common route string
!
dial-peer voice 201 voip
 no destination-pattern +91.T !not needed on an inbound dial peer
 no session target ipv4:xx.xx.xx.xx !not needed on an inbound dial peer
 no session transport tcp !as the outbound dial peer does not have TCP set it is not very likely that the inbound would need this
!
dial-peer voice 400 voip
 no session transport tcp
 no incoming called-number .T

!Apart from the changes to the dial peers you also need to turn on the Cube functionallity in the gateway for it to act as an SBC. Do that with this and also turn back on the security feature in the gateway.
voice service voip
 ip address trusted list
  ipv4 ITSP IP 1
  ipv4 ITSP IP 2
  ipv4 CM IP 1
  ipv4 CM IP 2
 address-hiding
 mode border-element license periodicity days 30

 



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Is there is any necessity to modify the gateway configuration

It would depend on your current configuration. I’ve previously suggested changes to your configuration, have you looked through these and incorporated any of them?



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Not yet, I'm thinking to modify only RL and RG configuration and keep the
gateway configuration as it is. As it would be easy for me, if any thing
goes wrong then I'll revert back the changes.

How hard can it be to revert back the configuration in the gateway? Off the top of my head I can come up multiple ways to do so. The advice given was based on errors/issues observed in the shared information. It’s up to you if you want to do it, no one forces you.



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Sincere apologies sir for asking such questions, you already helped me a lot by clearing all basics regarding MGCP, SIP, RL, RG, and so many more, but until this point I haven't gotten any chance to test things due to unfavorable production situation, all questions randomly pop-up in my mind considering xyz situations.

Now just want understand How CUCM will route calls to MGCP controlled PRI if my SIP Trunk is not in service, Browsed a lot but wouldn't find any relevant article or documentation which will assign priority between SIP Trunk and MGCP, I came across the "Ping Options" in the SIP profile but it would not help in call routing. 

That’s what Route List and Route Group are for. CM will use this construct to send calls to the secondary option if it fails on the first one.



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Hello Sir,

We have tested with RL & RG configuration the all works well but unable to
transfer calls MGCP PRI

I assume that you by “transfer calls MGCP PRI” mean send calls via the MGCP PRI? Are there any differences between how you need to send called and calling numbers between the two call paths? If so have you made any changes to the number transformation at the RL/RG level? It’s very likely that there is a difference between the two call paths and that the call is failing to work via the second path because of this. My advice would be to look at what you have configured for each path for a working setup or if you don’t have a working setup to create one per call path. Once you have that you should be able to combine them and set the appropriate configuration parameters to get it to work for the switchover.



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How do I check for the calling path and working setup for each

Look at the configuration you have in CM.



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Hii Kallberg Sir,

 

Yes you are right via MGCP PRI when the SIP PRI is not connected, And when I checked at the RL/RG level I didn't found any any number transformation there is only prefix digits are added. 

 

Now after making suggested changes at RL/RG levels and enabling cube functionality

 

For pattern 90

1st priority to => SIP TRUNK

2nd priority => MGCP PRI

 

When SIP PRI is up

 

CUCM - SIP TRUNK - GW 2951 - ITSP 

 

When PRI is down

 

CUCM unable to terminate call at any destination 

 

For pattern *90 

 

RL/RG Configured to route call via MGCP endpoint.

But when I changed the priority to route the calls via SIP trunk. 

 

CUCM - SIP TRUNK - GW 2951 - ITSP

 

The call is terminated with pilot number(022-4303-1000) not with DN numbers(022-4303-"4 digit DN)

 

But at the end we are unable to get the desired call flow after making changes, let me know what else details I can share with you or if you agree, can we connect over Google meet or anything else 

Please share screenshots of the route patters you reference and RL/RG level of these. Include the whole page for all of these.



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Hii Sir,

Here is the screenshots in PDF format with below link

Call Route Conf