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How CUCM/CME know which phone should accept a call

sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
[FONT=&quot]Hi guys, [/FONT]
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[FONT=&quot]I attached the diagram so that it would be easier for you to understand my question. There are two offices, A and B. A office has 7 495 7463449 number and office B has 1 312 9217816 number. Gateway, route patterns and dial peers are configured on both sides. [/FONT]
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[FONT=&quot]Now imagine someone from site A dials 1 312 9217816 or someone in site B dials 7 495 7463449. How CUCM or CME knows which phone (out a bunch of phones in each site) a cal should be forwarded to? [/FONT]
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[FONT=&quot]Thank you.[/FONT]
Best, sacredboy!

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    Legacy UserLegacy User Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
    Theres different ways this can be done. If they are dialing that one business number.One way You can have an auto attendant or operator answer the call and you dial the extension and it transfers to the called party. Or if you have a block of numbers purchased you can have an inward dial pattern like 7495741XXX and the last 3 digits match the extension.
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    negru_tudornegru_tudor Member Posts: 473 ■■■□□□□□□□
    you could assign a "secondary" number under the ephone-dn you want the call to go to.

    example:

    ephone-dn 101
    number 101 secondary 21555101

    this works if you had DID numbers for your stations otherwise, point the site's number to the front office lady or an auto attendant menu (as per the above post).

    EDIT: sorry, saw you mentioned CUCM - this would be valid for CUCME / CME.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
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    sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    dmarcisco, negru_tudor, thank you very much guys.
    Best, sacredboy!
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    sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    you could assign a "secondary" number under the ephone-dn you want the call to go to.

    example:

    ephone-dn 101
    number 101 secondary 21555101

    this works if you had DID numbers for your stations otherwise, point the site's number to the front office lady or an auto attendant menu (as per the above post).
    Under DID numbers you mean extensions. Is that correct?
    Best, sacredboy!
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    negru_tudornegru_tudor Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Under DID numbers you mean extensions. Is that correct?

    Well, as an example, if your extension is 7816, then your DID would be +1-312-921-7816. Final digits don't need to match necessarily but that's the way you'd do it. External callers would know they can dial straight to your extension using the above DID number. If you only have 1 DID/external number for the entire site / branch, then best point that to the nice lady @ reception and have her answer and transfer calls.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
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    Legacy UserLegacy User Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
    You know after answering this post I kinda miss voice lol
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    sacredboysacredboy Member Posts: 303 ■■■□□□□□□□
    Hi guys,

    Could you also clarify how exactly should the DID look like on CUCM and CME as i.e. should it contain country code and area code or it is PSTN's work to forward a call to the correct place?

    P.S. One little question. Is there any difference what number should be specified as primary and secondary on CME and line 1 and line 2 on CUCM.

    Thank you.
    Best, sacredboy!
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    negru_tudornegru_tudor Member Posts: 473 ■■■□□□□□□□
    lol tell me about it...got my CCNP Voice last year after 10-11 months of grinding for it and then got shifted to a server/application support project...can't tell you how frustrating that is - ended up forgetting a lot of the cool stuff I had picked up during my studies.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
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    negru_tudornegru_tudor Member Posts: 473 ■■■□□□□□□□
    sacredboy wrote: »
    Hi guys,

    Could you also clarify how exactly should the DID look like on CUCM and CME as i.e. should it contain country code and area code or it is PSTN's work to forward a call to the correct place?

    P.S. One little question. Is there any difference what number should be specified as primary and secondary on CME and line 1 and line 2 on CUCM.

    Thank you.

    OK, CUCM (server-based PBX) and CME (router-based PBX) are pretty different. In regards to the DID, the PSTN does all the routing and is responsible for handing the call over to your equipment so you don't need to worry about that too much. The DID number and how you configure it on the system (CME or CUCM) depends on how you draw up your dialplan and how the carrier delivers the dialed number to your site.

    For CME (router-based), you'd just do things like I posted above (ephone-dn 101 example). The key to the DID there is the "secondary" part - that would be your DID. So assuming you're in area code 21, exchange code 555 and the subscriber number is 101 (this is a bogus example by the way), the number you'd have to specify here under the "secondary" section is 21555101. You don't need to do the full PSTN number just (ie. +1-21-555-101), you might not even need to put the "21" in front..but that depends on how the carrier sends the number to you.

    For CUCM, things get a little more complicated because you'd have to get into "Called Party Transformations" and "Calling Party Transformations" which aren't that complicated but do require a little bit of reading up on. Here's a nice link that scratches upon these topics a little: https://supportforums.cisco.com/document/88291/cucm-digit-manipulation-call-flow
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
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    Legacy UserLegacy User Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
    lol tell me about it...got my CCNP Voice last year after 10-11 months of grinding for it and then got shifted to a server/application support project...can't tell you how frustrating that is - ended up forgetting a lot of the cool stuff I had picked up during my studies.

    Yea I hit a cross roads had 2 offers either go become the cisco Voice product expert over time at one place or become the infrastructure routing, switching, and data center guy at another place. If the voice track didn't become video heavy I think I would've taken the voice role. I was so turned off with where it was going I ended up taking the infrastructure role. Every time I go home I see my ccie collab lab catching dust and I just sigh lol
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    Legacy UserLegacy User Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
    Yes you can do it by the called/calling transformation pattern on CUCM but depending on your design you can offload that onto the voice gateway to do the the translating which is actually much easier. Then again it was easier for me since I came from deploying CME (where all the phone configuration is on the router) to migrating it to CUCM. It just made so much sense for me to have the CUBE router do all the translating when the call first comes in versus having the untranslated number passed to the CUCM to figure what to do with the digits.
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    MowMow Member Posts: 445 ■■■■□□□□□□
    dmarcisco wrote: »
    Every time I go home I see my ccie collab lab catching dust and I just sigh lol

    We should talk icon_wink.gif
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    Legacy UserLegacy User Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
    PM me if you're serious lol
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    MowMow Member Posts: 445 ■■■■□□□□□□
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    negru_tudornegru_tudor Member Posts: 473 ■■■□□□□□□□
    dmarcisco wrote: »
    Yes you can do it by the called/calling transformation pattern on CUCM but depending on your design you can offload that onto the voice gateway to do the the translating which is actually much easier. Then again it was easier for me since I came from deploying CME (where all the phone configuration is on the router) to migrating it to CUCM. It just made so much sense for me to have the CUBE router do all the translating when the call first comes in versus having the untranslated number passed to the CUCM to figure what to do with the digits.

    Absolutely, doing translation rules on the router instead of CUCM's Called/Calling Party Transformations would be waaaaay more convenient. Using Transformations gets useful when doing Globalized/Localized dialplans through CUCM. Thought posting these different ways of doing things for the sake of having CME's way vs. CUCM's way (overkill for single-site but useful for multi-site deployments).

    The video thing was a real turn off for me as well..might do the CIPTv2 one to get the Collab. cert as well (shame to let all that work go to waste) which could be a good chance to do a review of the NP Voice topics that mattered. On the other hand, network automation sparked my curiosity lately and I've been doing some Python courses to mix with some Cisco knowledge & was contemplating possibly starting studies for the CCNP R&S..though I'm not very sure I really want to dive into it just yet.
    2017-2018 goals:
    [X] CIPTV2 300-075
    [ ] SIP School SSCA
    [X] CCNP Switch 300-115 [X] CCNP Route 300-101 [X] CCNP Tshoot 300-135
    [ ] LPIC1-101 [ ] LPIC1-102 (wishful thinking)
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    Legacy UserLegacy User Unregistered / Not Logged In Posts: 0 ■□□□□□□□□□
    lol Yea I didn't get that far in my CCNP voice studies just ended up getting Cvoice and CIPT1 so never touched that topic Globalized/Localized dialplan so I personally don't know that perspective. I've only been exposed to CUCM single site designs with S2S vpns to connect small offices.

    Yea jumping into CCNP RS or any other track is much more satisfying if you are working with it. I kept putting off getting the CCNP RS since I was pushed and pulled to the be the RS, security, voice, wireless, and occasional DC guy so I was spread thin. Each project was lengthy competley different and theres no real overlap with some of the tracks and I ended up forgetting stuff from one track. It was a pain. I'm glad in new role I'm the RS/DC guy so theres overlap which makes my aptitude in the subject matter easier to retain.

    The idea of automation is awesome but haven't really found a way to apply it to my work that wouldn't make me have a panic attack when it's running and propagating across different devices. Only really automating I found useful as of yet is exporting csv files into any gui that I use to populate a large amount of ip's, mac addresses or something.
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