08-08-2016 01:33 AM - edited 03-18-2019 12:03 PM
Hi ,
I have 2 voice gateways have 2 pri's each.
my customer wants me to make a sip trunk pointing toward these voice gateways .
gateway will be h.323 as ISR 4331 does not supports MGCP.
I am planning to create 2 route groups and assign them in one route list.
I wanted to know if my pri 1 fails then call go through pri2 (here siptrunk1 will be used)
now when the pri 1 and pri 2 fails then how can redundancy be achieved that is how will call go through siptrunk2 to pri3 or pri4.
08-08-2016 04:23 AM
Hi,
Initially, this statement is contradicting.
"my customer wants me to make a sip trunk pointing toward these voice gateways .
gateway will be h.323 as ISR 4331 does not supports MGCP."
If you are creating SIP trunk then your gateway should be running SIP not H323.
Now, Route-Groups will failover between gateways but within the gateway dialpeers will manage failover between PRIs. Another way on the gateway is to create single trunk group which includes both PRI.
Make sure that the service parameter 'stop call routing on unallocated number' is set to false. Another way is to use this command on the gateways 'no dial-peer outbound status-check pots '
08-08-2016 07:29 AM
thankyou for your reply
we need to send calls from cucm to voice gateway via sip trunk, and we do not have mgcp so we will need to configure our gateway as h.323
about the paramters i have already thought over it and your statement matches mine here.
what if the 2 pri connected on one of my vg fails , will the call will be able to go to my second vg through another sip trunk which ill be adding in RL?
08-08-2016 08:45 AM
If your cucm is configured for sip trunk, your gateway should be sip. You don't need mgcp or h323.
If you apply the service parameter I mentioned to cucm or the command I mentioned in the voice gateway then the trunk will failover. Make sure to change the service parameter retry invite to 3 and timer trying to 150. This is important for proper failover to work
08-08-2016 10:15 AM
my gateway should be sip and i dont need mgcp or h323 . customer asking for this design.
I can use ICT trunk if i want to send calls to h.323
If i am using SIP then ill need to allow-connections sip to h323 and vice versa
also ill be needing 1 voip and pots dial-peer for both incoming and outgoing calls.
Stop routing services are for MGCP as I know. Please let me know if I am wrong,
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