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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

You have not provided the RL/RG details. Please open the route list details on each of the route groups that are included in the route lists and add that information to your document.

What is clear so far is this, on the route pattern used for sending calls to the PRI, ie *9.! you have a discard digit set to PreDot for the called number and your prefixing +226925 to the calling number. So this is what you need to put on the RL/RG level in the route list where you have the RG which holds your MGCP PRI as the second option so that your sending called and calling numbers in the format as your service provider expects it. Also on the other route pattern, the one used primarily for the SIP trunk, you are prefixing the calling number with +91224303. This as well needs to be put on the RL/RG level on the RG used for the SIP trunk as you cannot use the settings on the RP level if you want to achieve what you ask about.

On the MGCP PRI you’re using a CSS for called and calling number transformation named css-ISD. You need to look at what partition(s) that you have in that CSS and what transformation(s) that you have in these partition(s) as that would as well influence how you send these numbers to your service provider.



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Hello Sir,

I have implemented the recommended changes in the route list level for each priority wise added RG and same is attached in the below link starting from page 12, also attaching calling party transformation pattern.

Now as result when I assigned 1st priority to the MGCP (in my case RG_TATA_Wired_PRI), the call is terminated via MGCP, but when I assigned 1st Priority to the SIP, the call isn't terminated via MGCP when I SIP trunk unavailable

Call Route Conf2 

Sorry to say, but you need help with this as it’s evident that you don’t have the necessary skill set for this and without hands on assistance this would drag on for about forever. Please reach out to a reputable Cisco system integrator to acquire assistance with this.



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Good day, Sir.
I'm happy to report that all calls were terminated via MGCP yesterday when
I added a few missing partitions in the CSS-ISD and tested them by putting
a dummy destination IP in the SIP trunk configuration. I am grateful for
your assistance and thank you so much, sir, for being with me and answering
all of my questions; without this level of support, it would be impossible
for me.

Glad to hear that! However what you’re describing doesn’t make any sense at all as if the problem was caused by missing partition(s) in a calling search space calls via the route pattern that goes via the SIP trunk, ie the first option in the route list, would not have worked either.

Another thing, although not directly related to the problem you’ve been working on, by what you shared in the documents you have a CSS set for calling/called/redirecting number transformation. As that CSS, named “css-ISD”, doesn’t include any partition(s) used for transformation there is no need to have that set for transformation CSS’s on the MGCP PRIs. Recommend you to clear those settings and reset the MGCP trunks.

Add on bonus information, you should never ever mix call routing with transformations as that will lead to undesirable results. Any partition(s) and CSS(s) used for transformation should not include any selected partition(s) that holds configuration elements used for call routing, such as translation pattern(s), route pattern(s) or directory number(s).



Response Signature


Heartiest thanks to you sir 

You are correct that there is nothing related to the missing partition in CSS, The problem is in our testing method, in our setup the the SIP PRI is terminated on the Router (say Router A) - L3 core switch - Voice GW 2951 & the MGPC PRI is directly terminated on the Voice GW 2951, so in the testing we disconnecting Router A. Hence we are unable to route calls via MGCP PRI when SIP is unavailable.

Please use the correct terminology. A SIP trunk is not a PRI. That is an ISDN terminology and stands for Private Rate Interface. This has nothing to do with a SIP circuit.



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Hello Sir,

And also when I tested by assigning 1st priority to MGCP and 2nd priority to SIP trunk for the pattern 90 the calls isn't terminated to any destination 

Did you manage to get this sorted out?



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Yes sir, thank you so much for clearing that all my basic fundamentals, now
i have understood all the changes in the current configurations, since it
is in the production so I need to take downtime to implement it.

Regards,
Deepchandra Prajapati