We ordered from Telco E1 DID/DOD and I prepare my router to connect the e1 using e1 card. Bur the telco told us there is no availability now for e1 so that they provide us the DID using SIP trunk. and I do not understand how I can connect my callmanager to SIP using the DID/DOD.
Please if anybody know help me.
I did not receive any information till now form telco, shall I ask about any thing
I know it is silly question but this is first time I am facing this problem and it is not common in my country.
To my understanding, they should be providing you with an IP address of the termination point at their end, which you can configure your SIP trunk with in your call manager, and basically sending your PSTN traffic via SIP trunk to the provider. The rest would involve what type and how many digits they want to receive, based on which your call routing should be designed.
Read the links below about Cisco Unified border elements. after you look at some of the router configurations you should be able to ask your vendor /service provider the right questions to help you set up UBE. You are heading in the right direction. You shouldn't directly connect a sip trunk to your call manager simply becuse the CM GUI isn't roubust enough (command wise) to implement what you may need. And i am not too sure you would want to directly connect your phone system to the internet without a firewall in between. good luck. http://www.cisco.com/en/US/customer/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
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