Cisco CME Outbound Call taged to an extension

Posted on 2011-05-05
Last Modified: 2012-05-11
I have installed a 2911 ISR with CME and UE to run VOIP.  There are 20 phones however only 3 need to be able to make and receive calls.

There are 3 analog phone lines going into the system.  Creating a plar so that if someone dials phone number a, it goes directly to that specific extension was no problem so.  

555-1111 goes to extension 2000
555-2222 goes to extension 2001
555-3333 goes to extension 2002

My problem however is I cannot figure out how to make it so that if extension 2000 dials out, it only uses 555-1111.  Same for all 3 extensions.  All I could do was create a trunk group which I have named POTS.   I have assigned all three FXO ports to that trunk group so that when someone dials out, it picks a free line.

The users however want to ensure that the line associated to 2000 only gets used by 2000.  Same for 2001 and 2002 so how can I restrict it so that if 2000 dials out, it only dials out on 555-1111. etc.

Thanks for your help.

Question by:TheMetalicOne
    LVL 15

    Expert Comment

    I have not tested this but worth a try.

    Router> enable
    Router# configure terminal
    Router(config)# ephone 24
    your ephone number to which you want to make button specific call.
    Router(config-ephone)# auto-line 5 answer-incoming
    Router(config-ephone)# end  
    In this case when you select the line for a call a line selected that you specify in button

    LVL 79

    Expert Comment

    This irks me to no end... the whole idea of VoIP is the flexibility of outbound/inbound calls using mutliple lines and sharing POTS lines.... there is no simple correlation between a pair of wires and an IP phone like there is in the analog world. End users must be educated.

    Educate your users on the capabilites and educate yourself on the capabilities of the telco provider. They may be able to provide hunt-group features or other  options for you to explore.
    LVL 5

    Author Comment

    Deepdraw: thank you for your useful response, I will try it out.

    Irmore: Thank you for your insightful and completely useless comment in respect of the actual question.

    The analog lines are temporary until the Sip trunk will be in place. In the meantime this is how they want to manage it.

    What irks me to no end is when someone determines their time is best spent offering opinion and insult instead of actually answering the actual question. If you don't have the direct answer to the question, you are more than welcome to find a forum on another question as I am sure there are others that suit your way of doing things better.

    If there was only one way of doing things, we wouldn't have options. Strange huh?
    LVL 79

    Accepted Solution

    TheMetalicOne, my humble appologies for apparent rude behavior. It was not my intent, I promise.

    Nobody likes a challenge like I do, that's why I participate on EE.
    All that said, there is still not a simple method to do what you are wanting to do.
    Inbound calls from a specific POTS line to a specific internal DN is not a problem, as you have done with a simple PLAR. I don't think the auto-line selection feature is going to help you at all either.

    Outbound calling is vastly different because we create dial-peers to match dialed number patterns and send that called pattern to a specific analog port, or group of ports, and generaly assign preference if more than one egress port.
    I've been doing Cisco voice for a long time and have never come across a way to designate a specific ephone-dn to be forced out a specific analog port.
    With a PRI, we can use translation patterns to make sure outbound calls from a specific DN goes out as a specific DiD number. Not so with analot POTS lines because the number is stuck to the pair of wires at the telco CO, and cannot be over-ridden by a tanslation pattern.

    Let me play around with CME in my lab and maybe I can come up with something.

    LVL 5

    Author Comment

    I have informed the users that this is simply not the way the voip system works, and indeed they have to live with the random line output until they get a PRI or SIP trunk in place.

    Problem solved by the route of acceptance.

    Cheers, thanks for trying.
    LVL 5

    Author Closing Comment

    The solution is to simply not do it this way.  Thanks

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