01-11-2023 07:05 AM
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
Solved! Go to Solution.
07-02-2023 10:47 PM
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.
07-03-2023 02:11 AM - edited 07-04-2023 12:13 AM
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.
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
07-24-2023 12:03 AM
Is there is any necessity to modify the gateway configuration
07-24-2023 01:57 AM
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?
07-24-2023 06:32 AM
07-24-2023 06:59 AM
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.
07-25-2023 11:28 PM
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.
07-26-2023 01:21 AM
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.
07-28-2023 11:04 PM
07-29-2023 12:00 AM
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.
07-29-2023 12:44 AM
How do I check for the calling path and working setup for each
07-29-2023 01:11 AM
Look at the configuration you have in CM.
07-29-2023 10:38 PM
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
07-29-2023 11:34 PM
Please share screenshots of the route patters you reference and RL/RG level of these. Include the whole page for all of these.
07-30-2023 01:26 AM
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