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Multiple Vendor (Provider) SIP Trunks with 1 CUBE

B_VOIP_NE
Level 1
Level 1

Hello,

We are looking into migrating our SIP provider from one vendor to another.  I have an open interface on our CUBE and would like to connect the new provider here.  We will be migrating off location by location. We have about 10 different DID blocks for the different locations which all come back to this CUBE for PSTN access.  Is there a clean way to do this with dial-peers?  I would like to set it up so anything with this prefix "780-XXXX" goes out the new dial-peer for the new interface and new provider.  Any other existing DID's continue to go in and out the old vendor.  As we migrate more sites/DID blocks over I would like to be able to add these to this dial-peer to use the new provider.  Is this possible at the dial-peer level or would someone have a better way to do this?

 

We have about 10 sites with 2 vendor SIP trunks at our 2 main sites.  We have a CUBE at each location, a publisher at our primary location and a subscriber at our secondary location.  The sites are divided between the 2 data centers (half go out one site the other half out the other controlled at the device pool "Standard Local Route Group" level.

 

The vendor is also going to provide a set of new "test" numbers that I would add to this dial-peer first so I can use this to test connectivity out/in.

 

Can someone let me know if this is possible and how I would do this?  Current dial-peer's for inbound/inbound are below.

 

dial-peer voice 1 voip
 description Incoming calls from provider
 translation-profile incoming Strip-AREACODE
 session protocol sipv2
 session target sip-server
 incoming called-number .T
 voice-class codec 1
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax-relay sg3-to-g3
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 no vad

dial-peer voice 2 voip
 description Outbound calls to provider
 translation-profile outgoing Strip-9
 destination-pattern 9T
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax-relay sg3-to-g3
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
 ip qos dscp af32 signaling
 no vad
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