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New Member

change pstn number in voice gateway

Hi all,

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.

5 REPLIES

Your voice gateway registers

Your voice gateway registers via MGCP? Not sure what PSTN number you are referring to changing? Do you have a single number that every phone uses when you dial out and you would like to change that?

New Member

Here is my voice router

Here is my voice router config:

 

voice service voip
 ip address trusted list
  ipv4 10.16.1.20
  ipv4 10.21.1.5
 allow-connections h323 to h323
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 h323
  call preserve limit-media-detection

!         
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729br8
 codec preference 3 g729r8
!
voice class h323 1
  h225 timeout tcp establish 3
!
!
!
!
voice translation-rule 1
 rule 1 /^46\(..\)/ /132\1/
!
voice translation-rule 2
 rule 1 /^746\(..\)/ /132\1/
!
voice translation-rule 3
 rule 1 /^0746\(..\)/ /132\1/
!
voice translation-rule 4
 rule 1 /^20746\(..\)/ /132\1/
!
!
voice translation-profile profile1
 translate called 1
!
voice translation-profile profile2
 translate called 2
!
voice translation-profile profile3
 translate called 3
!
voice translation-profile profile4
 translate called 4
!

controller T1 0/0/0
 cablelength long 0db
 channel-group 0 timeslots 1-24
!
controller T1 0/0/1
 cablelength long 0db
 channel-group 0 timeslots 1-24
!
controller T1 0/1/0
 cablelength long 0db
 channel-group 0 timeslots 1-24
!
controller T1 0/1/1
 cablelength long 0db
 channel-group 0 timeslots 1-24
!
controller T1 0/2/0
 cablelength long 0db
 ds0-group 0 timeslots 1-11 type e&m-wink-start
!

dial-peer voice 1 pots
 translation-profile incoming profile1
 shutdown
 incoming called-number 46..
 direct-inward-dial
!
dial-peer voice 2 pots
 translation-profile incoming profile2
 incoming called-number 746..
 direct-inward-dial
!
dial-peer voice 3 pots
 translation-profile incoming profile3
 incoming called-number 0746..
 direct-inward-dial
!
dial-peer voice 4 pots
 translation-profile incoming profile4
 incoming called-number 20746..
 direct-inward-dial
!
dial-peer voice 5 voip
 description -= COL-UC-VCM1 =-
 preference 1
 destination-pattern 132..
 session target ipv4:172.16.1.20
 incoming called-number .
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 6 voip
 description -= MIL-UC-VCM1 =-
 huntstop
 preference 2
 destination-pattern 132..
 session target ipv4:172.21.1.5
 voice-class h323 1
 dtmf-relay cisco-rtp h245-signal h245-alphanumeric
 no vad
!
dial-peer voice 10 pots
 description -= Local Calls =-
 translation-profile incoming profile1
 destination-pattern [2-9]......
 port 0/2/0:0
 forward-digits all
!
dial-peer voice 11 pots
 description -= Long Distance Calls =-
 destination-pattern 1[2-9].........
 port 0/2/0:0
 forward-digits all
!
dial-peer voice 12 pots
 description -= International Calls =-
 destination-pattern 011T
 port 0/2/0:0
 forward-digits all
!
dial-peer voice 13 pots
 description -= Info Routes =-
 destination-pattern [2-7]11
 port 0/2/0:0
 forward-digits all
!
dial-peer voice 14 pots
 description -= Emergency Routes =-
 destination-pattern 911
 port 0/2/0:0
 forward-digits all
!
dial-peer voice 7 pots
 incoming called-number .
 port 0/2/0:0
!
!
gateway 
 media-inactivity-criteria all
 timer media-inactive 5
 timer receive-rtp 1200
!

Hope not too much and please advise what commands to be changed ?

Thank you very much.

Without knowing all elements

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.

New Member

Hi Michael, Thank you for

Hi Michael,

 

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.

Thank you.

You may be able to go to your

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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