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Route DID to Freepbx setup

Wonxie
Level 1
Level 1

Hi,

I have below setup.

ISP PRI >> Cisco Voice Router >> CUCM >> FreePBX.

All my did are on Cisco voice router and cucm is connected to voice router via h323. my cucm is using 4 digit dialing. All calls to the pstn are passed by cucm to voice router and user has to use 9 as access code for outside line and the voice router manipulates it to did as per isp requirments.  similarly on incoming calls from pstn 4 digit number is passed to cucm which matches last 4 digits of dialed number. 

 

Now i have Freepbx at remote site and it has its own set of 3 digit dialing.A sip trunk is in place between cucm and freepbx and calling is working fine between these sites.

Now i want to map a DID to those 3 digit numbers registered on phones behind Freepbx for incoming and outgoing calling. that remtoe site has access code of 4 when reaching it from cucm registered phone. and that site is not using any access code to reach the extensions behind cucm.

 

For incoming calls form PSTN i have translation pattern which translates the called number to 4.xxx where xxx is specific extension and rmeote site behind freepbx. this is working fine. and for each DID i have to put a translation pattern as the numbers at freepbx site are not in sequnce and not all numbers behind freepbx are assigned a DID. 

 

I am trying to figure out how to make all outgoing calls from freepbx to the pstn via the cucm route correctly using proper outgoing DID ?

What can i use in cucm that when i recive a call to pstn from 3 digit ext behind freepbx being translated to a 4 digit number as my router will do remaining pstn formating. I just need to convert the 3 digit number behind the freepbx to a 4 digit number in the numbers range assinged to us by isp.

Thanks a lot for guidance.

2 Replies 2

Daniel Bosch
Level 1
Level 1

In CUCM, you can use Calling Party Transformation Patterns to accomplish this.  You'll need to set up a partition that will contain the patterns.  You also need to set up a CSS with the new partition as a member.  The CSS will be applied to the SIP trunk between CUCM and FreePBX under Outbound Calls > Calling Party Transformation CSS.  You will also need to uncheck the "Use Device Pool Calling Party Transformation CSS" box and reset the trunk.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/CUCM_BK_C95ABA82_00_admin-guide-100/CUCM_BK_C95ABA82_00_admin-guide-100_chapter_0110001.html

Hi,

The other option is to apply the 4 digits transformation on the Freepbx and adjust the intercluster call routing on CUCM so no other translation is needed.

 

HTH

 

Regards

 

Carlo

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