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Splitting of SIP trunk and translation Rules on voice gateway

Spartan
Level 1
Level 1

Hi Folks,

well, i have SIP trunk having 100 Calls DID  & all are being used for Company A.  those are configured on VG & working perfectly fine . I have to reserved 20 DID out of 100 for Company B  . Can someone give hint  how to do that.  So any call from external to first 80 DID will goes to Company A while other DID call will goes to company B .

7 Replies 7

The second company has different UCM cluster?  is this Two cluster sharing one voice gateway ?

 

 

 

 



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Nope, the Company B will use the same UCM which Company A is using , both will have the same gateway as well. 

Then there is nothing to do in the gateway. You would do the needed configuration to split the numbers up in CUCM. Put the directory numbers of company A into one partition and the directory numbers of company B into another partition. These two partitions need to be in the calling search space that is used on the SIP trunk between the CM and the SBC gateway.



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Assume, your DID block as 24588600 - 24588699. Company A use DID range from 24588600-24588679 And company B 24588680-2458899.

 

I am using 4 digit extensions range for my phone.For company A, extensions will be 8600-8679. And for company B 8680-8699.

 

Company A.

  • Created  partition P_compA
  • Created   calling search space CSS_A with partition P_compA
  • Assigned partition P_compA for extension range 8600-8679

 

 

Company B.

  • Created  partition P_compB
  • Created   calling search space CSS_A with partition P_compB
  • Assigned partition P_compB for extension range 8680-8699

 

For gateway to call these partitions.

  • Created   calling search space CSS_Trunk with partition P_compA & P_compB
  • Assigned inbound calling search space CSS_Trunk on trunk.

 

 



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As I’m a big advocate of E.164 I would recommend to do this instead of piping down the directory numbers to four digits.

Assume, your DID block is +4624588600 - +4624588699. Company A use DID range from +4624588600-+4624588679 and company B +4624588680-+462458899. I’ve used +46 in the example since I’m based in Sweden.

As most of the users are likely to be lazy and don’t want to dial the full numbers on their desk phones you’ll add a layer of translation pattern, on top so to say, to cater for dialling in the the short number format. For users on software clients, like Webex, or Jabber if this is still used, this should not really be an issue as for the most they would likely dial via information provided in the directory, aka they would actually not enter any digits manually.



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well that great to hear . I think little confusion. I have Sip trunk having  capacity of 100  channel or call/Numbers . I want to define on VG if call  coming from   (  External to VG) 24588600-24588679  . the call will go to handler (CUC) and company A Call Handler   recording will run while when the Call come to rest 20 Cannels, The Company B  Call handler handler will run. I have define the call handler with Virtual DN & its fine. i just need support to do diel peers /  Translation rule  ( IN ward  & outgoing calls ) to define on VG  . 

Since you did not mention CUC in any previous post there is no chance of anyone knowing that this would be part of the system landscape.

Nevertheless still not something that you would need to do in the VG. If you want the call to first go to a call handler in CUC just do a translation pattern in CM that matches the number range of company A and another that matches the number range of company B. In the TPs you translate the called number into the numbers you have set on the call handlers.

If you anyway absolutely want to do this in the VG have a look at E.164 number maps to have individual dial peers to the CM per company.

Please have a look at this excellent document for details. https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html



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