CME and Multiple "Lines"

[mattv123]

Hey guys,

I have my Cisco CME 4.0 working great with one numbers, but I need to be able to support at least 2 more lines for both incoming and outgoing calls.

So far all providers say I need to have separate accounts (aka sip-uas) to have access to multiple numbers. In CME i am restricted to one.

I had found a hack where I believe the guy tricked the system into authenticating under a pots peer, but without that hardware, I cannot make that work.

Any suggestions? I would love to get this working ASAP, and maybe there is a solution that does not require multiple SIP accounts but runs over one SIP trunk?

Any help is greatly appreciated in advance.

Thanks!

[openeyes]

Hi,

Your right you can only have one sip-ua and I am not aware of any real way to hack that.

However you do have a few options. If you have an account with a VSP with say three numbers each with different username/passwords that you can have the one sip-ua and authenticate the username (and thus the phone number) at the dial-peer level. That way you can have a several numbers.

If you have two different VSP's then you will need a SIP proxy/registrar server like ser http://www.iptel.org/ser/. With this you can authenticate the CME with the ser and the ser can have a number of SIP trunks ie redirect your calls. I am just in the process of setting up a test.

Hope that helps.
openeyes

[mattv123]

Thanks for the help! Well I have actually pursued your former recommendation.

The problem I am having now pertains to getting our service provider to support Cisco's CME voicemail setup. To keep it simple, since Cisco is so stringent on standards, CME makes the VSP responsible for transferring calls that are being routed from the auto attendant or to voice mail. IT does this through REDIRECT and MOVED TEMPORARILY SIP headers. My VSP got the redirect part to work, but out voice mail doesn't work because they can't seem to get the Diversion header that is required to identify who was originally being called before a call was routed to voice mail. So, voice mail doesn't work- kinda sucks.

Oh well, once that is going we will be good. Hopefully my VSP will be able to fix this issue, otherwise I will need to find another. Any recommendations?

[krwalczak]

I have the very same issue.....

I can get all the numbers registered to the providers registrar server using the "credentials username 1234 password 1234 realm test.com" command for each account/number in sip-ua.

This gets the inbound calling to come in and hit the correct ephone-dn entry matching the number (with a no-reg both since registration is at the global sip-ua level).

Problem is outbound calling. Provider still needs the Digest Authentication for the number calling out. That is handled by the authentication command in sip-ua. But, can only be for 1 account/number. So, any number other then the one with the correct authentication account info fails to Auth with the SIP server for outbound dialing.

I've tried manually creating POTS dial-peers for each number. Since POTS dial-peers are the only one that can hold individual Authentication info. I just can not get those manually creted dial-peers to link to the dynamic ephone-dn created dial-peers.....

If anyone has a sample config I can look at for dial-peers done this way but linking/working with ephone entries. Not a physical voice (POTS) port. It would be greatly appreciated.....