CCME express offnet call

Unanswered Question
Jul 28th, 2006
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I have 2 sites with CCME in both. both routers have FXO ports connected to the PSTN. How can i setup the system to be able to have the capability to call from the local IP phones of Site-A to the PSTN of Site-B? is this possible? how can i map the dial peer to the FXO of the other site?

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Brandon Buffin Fri, 07/28/2006 - 05:18
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If you wanted to send calls from SiteA to area code 408 out the PSTN connection at SiteB, you would need a configuration such as:


SiteA

dial-peer voice 1 voip

destination-pattern 91408.......

session-target 10.1.1.1


SiteB

voice service voip

allow-connection h323 to h323


This is not a complete configuration, only the relevant additional commands needed in this scenario. This assumes that SiteB has a working dial peer that will match the last 7 digits of the number represented by ....... as the 91408 will be stripped by default. You can use the forward-digits command to specify the number of digits to forward if you do not want all explicitly matched digits to be stripped.


Hope this helps. If so, please rate the post.


Brandon

a.gooding Fri, 07/28/2006 - 05:32
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secnario: SITE A places a call and should recognize digits as a pstn call that should go to SITE b FXO. if site B however has all ports in use, that same call should be able to come back and be placed on SITE A local FXO ports.

(Note: you have to plan your dial-patterns accordingly )

so suppose i place a call to 868XXXXXXX and that is a number that will save cost if it goes through SITE B.


therefore router A will have two dial-peers, POTS and VOIP


dial-peer voice 1 pots

destination-pattern 868xxxxxxx

port 1/0/0

no vad

no digit-strip

preference 2


dial-peer voice 2 voip

destination-pattern 868xxxxxxx

session : remote CME

dtmf-relay


Now, on router B we will have one pots dial-peer matching pattern


dial-peer voice 1 pots

destination-pattern 868xxxxxxx

port 1/0/0

no digit-strip

no vad


the default preference for this pot is 0 which is highest. therefore, the call should go through from router A using the VOIP then match the POTS peer on B and if for some reason the associated port is busy, it should revert to the orginal FXO at site A

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