How Do You Integrate CUCM 6.XX and Asterisk?

Discussion in 'General' started by coolguythampy, Oct 11, 2010.

  1. coolguythampy

    Joined:
    Oct 11, 2010
    Messages:
    5
    Likes Received:
    0
    Hello,


    I am interested to know whether we can integrate CUCM and Asterisk seamlessly. My office uses Elastix as VoIP solution and all other offices outside my country (same firm) uses CUCM.


    Elastix version used: 1.6 with Asterisk 1.4 integrated into it.

    CUCM version Used: 6.xx


    We have Polycom, thomson, some linksys and CISCO phones provisioned in our office. We would like to integrate the CUCM in other offices to our Elastix server.


    I have read on forums about 2 different methods to connect these two.


    1. Using SIP

    2. Using H323


    We do not have any gateway concept in our office right now (For that purpose, we use a thir party for bridging functionalities. We will be shortly buying gateway for our own use).


    The way we read in forums, and blogs it seems the communication from CUCM to Asterisk is fine, but from asterisk to CUCM is not proper.


    Anyone has any experience regarding this one? What is your take on this? We might be moving to CUCM V 8.XX shortly.


    Also, does CUBE come into play in CUCM and Asterisk integration?
     
  2. coolguythampy

    Joined:
    Oct 11, 2010
    Messages:
    5
    Likes Received:
    0
    Anybody has any idea regarding this?
     
  3. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
  4. coolguythampy

    Joined:
    Oct 11, 2010
    Messages:
    5
    Likes Received:
    0
    Hello,

    Thanks for your "help", but I did do that. The reason I posted here is that I am a newboe to this. So, I get lot of conflicting advice in various forums and wikis. Some say this works while others say they dont work.

    I heard SIP trunks work buit we need some device like cube to make the integration smooth. But some say it's not efficient and would not be the choice and should go for h323.

    Asterisk, i think has oh323 and some say it has problems in connecting from Asterisk to CUCM.

    So with lot of conflicting advice, I was looking for a solution from people who has really implemented them.

    I read soem useful info on trixbox forum, but not sure if it applies to our elastix build used
     
  5. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    So after you actually read

    http://www.voip-info.org/wiki/view/Aste ... ntegration

    which part did you not find helpful? It seemed to me to explain exactly how to connect Asterisk(Elastix) with your cisco stuff, it covered it all the way to specifically Cisco Call Manager 6.1. which I believe is you, no?

    I really suggest you actually read the stuff you are presented with and THAT you should do sincerely.

    After a few hours study you might reconsider your possibly sarky '. . . Thanks for your "help" . . .' reply, maybe in your case (possibly ADD, have you been tested?) you will need to read it twice, maybe more for full retention.

    If you are not up to that challenge, or simply didn't like my help, then the link at the top of this page for "paid support" from PalaSanto will, I'm sure, fix you up.
     
  6. coolguythampy

    Joined:
    Oct 11, 2010
    Messages:
    5
    Likes Received:
    0
    Hello,

    I did not mean to offend you or anything. Just that read too many articles on integration and all getting mixed up.

    I'm sorry if I sounded a bit rude, but I did read the wiki. Since I am completely new to all these, it's not very clear to me and when I read other forums, it seemed to be conflicting.

    Can you help me out with the question as to wherther CUBE (Cisco Unified Border Element) device is needed to this to work?

    Our Elastix servers are configured with IAX trunks. Not sure if seemeless interoperability is possible
     
  7. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    It is almost impossible to offend me :), no apologies necessary.

    I'm sorry that you are confused and conflicted by other advice, I just ask you "have you tried the voip-info recipes" yet?, They seem very explicit as to how to talk to the Cisco stuff. If you aren't prepared to try it , how will you ever know?. Please respond AFTER you followed (not just read) the posting. I can however assure you that Cisco stuff does not talk IAX2 (Inter Asterisk eXchange Protocol) you will need to use SIP (easily) or the more legacy h323 (much harder), Both are covered in my reference.
     
  8. coolguythampy

    Joined:
    Oct 11, 2010
    Messages:
    5
    Likes Received:
    0
    The reason i spend so much time on making sure them methods/theories work is because we have only one test lab. Most of the time the network team will occupy it and so it will be difficult to get a booking. So I needed to make the maximum use of the time I get at the test lab. Ok, I wilol try it out first and then come up with any questions I have
     
  9. garcia.ronald.d

    Joined:
    Sep 24, 2008
    Messages:
    134
    Likes Received:
    0
    Hello Dicko,

    Im trying to integrate a CUCM and Asterisk, I followed all the instructions in the web but I dont have a positive result.

    I followed the instructions in http://www.techieanalyst.net/howto/cucm ... fault.aspx

    and

    http://www.voip-info.org/wiki/view/Aste ... ntegration

    and always Asterisk show:

    -- Called callman/83001
    -- SIP/callman-00000090 is circuit-busy

    I´ve read that need a MTP but the CUCM have yet a MTP that point to itself.

    I could use a GW Cisco E1 as a gateway pstn for asterisk, but Call Manager-Asterisk, its break my head.

    Please if you can help me?
     
  10. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    I have only done a couple of these integrations, and they have all been interim solutions as they were replaced (I don't like paying for stuff that doesn't work very well) so right now I have no way of verifying , my memory was that it just kind of worked following the voip-info thread, and getting the authority mechanism and user/password to agree.

    sorry I can't be more helpful.
     

Share This Page