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-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 in my case if customer A send me the ANI as 33222222 which is for > customer B then the call will hit dial-peer 2 and the call will go outside > with 33222222 as the ANI number so how we can block Site A calls if they > didn't use there own DID. > > > > > _______________________________________________ > For more information regarding industry leading CCIE Lab training, please > visit http://www.ipexpert.com/ > > Are you a CCNP or CCIE and looking for a job? Check out > http://www.platinumplacement.com/ > > > > > _______________________________________________ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > Are you a CCNP or CCIE and looking for a job? Check out > www.PlatinumPlacement.com >
_______________________________________________ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
