Did you configure the cucm server up
Address on the cme?

Randall



On Sep 14, 2012, at 7:32 PM, Bill Lake <[email protected]> wrote:

> Because they are coming in on a seperate sip dial peer, but maybe that won't 
> work.  I would think it wouldo because you would only apply the change 
> inbound from them ( you said they are changing ani in their switch) and only 
> allow the ani you want
> 
> On Friday, September 14, 2012, John John wrote:
> I get what you are talking about but this is not my case.
>  
> let me explain again:
>  
> we have one voice gateway with 2 E1 for 2 customers A,B
> and the connection between us and customer A,B is sip trunk so they have 
> their own PBX and I have already assigned range of DID for both company and 
> as you know I have 2 dial-peer between me and the customer PBX.
>  
> dial-peer voice 1 voip
> description ## Customer-A ##
> answer-address 332211..
> destination-pattern 332211..
> modem passthrough nse codec g711ulaw
> session protocol sipv2
> session target ipv4:192.168.10.10
> dial-peer voice 2 voip
> description ## Customer-B ##
> answer-address 332222..
> destination-pattern 332222..
> modem passthrough nse codec g711ulaw
> session protocol sipv2
> session target ipv4:192.168.20.10
>  
> so now for ougoing calls from customer A the call path will be   Customer A 
> PBX---------Mygateway------PSTN over E1
> so If customer A send the calling number as 33222222 which is customer B 
> range then the call will go with customer B DID so how can I block them.
> 
> From: Bill Lake <[email protected]>
> To: John John <[email protected]> 
> Cc: Pavan <[email protected]>; "[email protected]" 
> <[email protected]> 
> Sent: Saturday, September 15, 2012 1:11 AM
> Subject: Re: [OSL | CCIE_Voice] sip trunk
> 
> Sure you can, you know that any call incoming from Site A must have ANI of 
> 332211 and any call incoming from Site B must have 332222 so you just need to 
> enforce that and you could do that with a incoming voice translation rule.  
> 
> So no matter what the customer sends you can change that to what you want it 
> to be
>  
> We do this all the time in the work books
> 
> voice translation-rule 1 
> rule 1 /^3...$/ /408387\0/ 
> 
> this expands 3001 to 4083873001 
> 
> but you could also do
> 
> rule 1 /^3...$/ /4001/  
> 
> This would take any 3XXX number and change it to 4001, why you ask, just 
> think you have 800 people calling out from your company call center and this 
> prevents 800 different numbers from going out. Everyone you call gets 4001.  
> not 3001-3799 or something like that.
> 
> You get the idea?
> 
> 
> 
> On Fri, Sep 14, 2012 at 4:35 PM, John John <[email protected]> wrote:
> Dear Pavan,
>  
> Thanks for your feedback,But my question is even if i implement cor still I 
> can't control that because I can't control what customer A and B and in that 
> case if customer A sent me the call with ANI for cutomer B then customer B 
> dial-peer will be match and the call will not be blocked.
>  
> but during my google search I found some greating feature which is incoming 
> uri based on the IP so anybody working in that feature ????
> 
>  
> From: Pavan <[email protected]>
> To: John John <[email protected]> 
> Cc: "[email protected]" <[email protected]> 
> Sent: Friday, September 14, 2012 6:43 AM
> Subject: Re: [OSL | CCIE_Voice] sip trunk
> 
> I think cor can offer a good solution for this. Assign a separate cor group 
> to each dialpeer say cust1 and cust2 in both inbound and outbound directions
> 
> -Pavan
> 
> On Sep 13, 2012, at 17:45, John John <[email protected]> wrote:
> 
>> Dear All,
>>  
>>  We have two sipt trunk for 2 comapany:-
>>  
>> company A - DID range 332211XX
>> Company B - DID range 332222XX
>>  
>> and each company has own PBX,Company A  has AVAYA and company B has cisco 
>> Call manager.
>>  
>> and they have sip trunk to my gateway where is the E1 is connected.
>>  
>> in my gateway there is two dial peer :
>>  
>> dial-peer voice 1 voip
>>  description ## Customer-A ##
>>  answer-address 332211..
>>  destination-patt
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