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Dial-peer Statements

I want to setup a gateway router between PSTN and PBX with 5 Pri?s between the PSTN and the gateway router, with 5 Pri?s connecting from the Gateway Router to the PBX. At first the calls will all be routed to the PBX. Users will be migrated from the PBX to CCM over a two month period.

If a user migrated off the PBX to CCM, will I be able to use a VoIP dial-peer on the gateway router to reach this user from the PSTN. Will the VoIP dial-peer statement send the call directly to the CCM, if called from the PSTN?

Example:

dial-peer voice 1 voip

destination-pattern 3601234 ? (user extension that moved to CCM from PBX)

session target ipv4:10.10.10.1 ? (CCM IP Address)

2 REPLIES
Blue

Re: Dial-peer Statements

if CCM is controlling the PRIs that connect to the PSTN, and you have the gateway configured correctly, when a call comes in that belongs to a CCM user then CCM will see this and route the call straight to the phone, without the need for a VoIP dial peer.

a voip dial peer would be required if a call came in from the PSTN to the PBX and the user was a CCM user. in this case, the PBX would need to send the call to the CCM via a voip dial peer.

please see the following links for more info on dial peers and their operation:

understanding dial peer status:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010e6d1.shtml

understanding dial peer call legs: (linked from first document)

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml

understanding inbound/outbound dial peers: (linked from first document)

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml

Blue

Re: Dial-peer Statements

This will work much better if you use MGCP instead of H323. The Call Manager will route the full range of extensions to the PBX, and as IP phones are added the call routing will work with no additional route patterns or router config.

The problem with using H323 is that the router can route calls without the Call Manager. So if you are receiving digits 1XXX from the PSTN, and want to send those calls through the Call Manager you will need a VOIP peer pointing at the Call Manager, and some sort of bogus number range for the POTS peer that points to the PBX with a translation to send the correct digits to the PBX. Otherwise the calls will just route directly to the POTS peer and never pass to the Call Manager. Your solution would work, but would require you to add a peer for each extension you move.

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