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dial-peer and call routing


in the attach I've put scheme of my network and also configuration of my Gateway and CME. From outside I can reach all phones (IP and analog phones). User behind PBX dial 0 (zero) when they want call any outside number as well dial IP phones. IP phones dial any number without any prefix. Problem are: when PBX users dial some outside number which start as local numbers for example 214... they reach locals on PBX, also when they dial short code (122) they got local. PBX is a "black box" and we cannot do anything with it. So is there any way to do some configuration on a gateway or CME to solve problems.

My questions are:

1. Must I have dial-peer for any phones behind PBX on "port 0/2/0:15"?

2. Problem with transfer call on IP phones. When somebody got a call from outside and want to transfer to other Ip phone the call went out!? I suppose it is becouse of "dialplan-pattern 1 xx286... extension-length 3". But how can I solve that IP phones when call outside has their CLID.


Re: dial-peer and call routing

first u dont need a dial-peer for every IP phone extntion number

u can use one that include all thorugh a wildcard like

dial-peer voice 28 voip

destination-pattern [1,2,3]..

session target ipv4:

i see u have trnaslation profiles like

oice translation-rule 22

rule 1 /^711610/ /138/

rule 2 /^711611/ /135/

are pbx users dial 711610 to get to 138?

is here where u have problem with conflict with outsude calls ?

for configlctin with ip phones numbers and outside number u can do a trick like

let the pbx users dial a su have 0 then 8 then the ip phone number

if u wanna cal number 139 they call 08139

0 will be striped on the pbx as u have it on the gateway u only need to strip the 8 from the number befor send it to ip phones u can do on the CME

now on ur gateway u will have the dial-peer like

dial-peer voice 91 voip

destination-pattern 8[1,2,3]..

session target ipv4:

on CME

voice translation-rule 1

rule 1 /8\(...\)/ /\1/

voice translation-profile to_cme

translate called 1


dial-peer voice 28 voip

incoming called-number 8...

translation-profile incoming to_cme

the above config in cme will take any call consisit of 4 digits and start with 8 the trnaslation profile will strip out the 8 and the call will go to the ip phone normally

good luck

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