Please help with the following question: we have a voice gateway to reach local PSTN and this gateway registers to CUCM in HQ. Recently we need to change the local number for PSTN dial, what should we change in the voice gateway config ? any change in CUCM config ?
Thanks for all comments.
Actually the new line is Flex-T1 so I need to config SIP Trunk. Please advise.
Without knowing all elements it would be difficult for me to give you an exact list of commands to use for this scenario. Let's just say your ITSP will expect the same digit presentation as they were with the PRI. You could pretty much do a direct copy/paste of your outbound dial-peers. For example
dial-peer voice 10 voip (changed from pots) description -= Local Calls =- translation-profile incoming profile1 destination-pattern [2-9]...... session target ipv4:(ITSP IP address)
session protocol sipv2 forward-digits all
You should also add under voice service voip
allow-connections sip to sip and possibly your ITSP to the trusted list.
Also I would consider creating a SIP trunk between the gateway and CM and remove H323 once you have phased out your older circuits.
Thank you for jumping in to advise. Actually I already have a range of PSTN numbers being used for dialing out to PSTN but recently the service provider wants to upgrade this service to SIP trunk so I believe I need to change the configs in my voice router too - currently it has been registered as H323 gateway.
You may be able to go to your dial-peer configuration that was pointing to your PSTN and add session target ipv4:x.x.x.x or session target dns:domain.com (dns required to be configured on your router). Then add session protocol sipv2 and this will establish a SIP trunk to your provider using your existing dial-peers. At that point I would look at your voice service voip configuration and make sure allow-connections sip to sip | sip to h323 | h323 to sip | h323 to h323 is configured. Depending on model number you should look at ip address trusted authenticate command as well.
Or you could turn it into a SIP gateway and basically duplicate your dial peers. Have your provider allow both "circuits" to be up while you migrate over.
You may run into + dialing issues in the future if you are trying to send + from CUCM to your H.323 GW so you may want to go ahead and change it to SIP.
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