Help ..no trunk online

Discussion in 'General' started by waqasbhutta, Jun 1, 2009.

  1. waqasbhutta

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    hi there

    i am having a strange problem

    all my trunks are not getting registered
    neither any extension are working
    all i did yum update
    and now nothing works
    everything looks fine in the pbx
    all i see request sent or unreachable

    i checked the cable connection looks fine as i can access it through my internal ip
    any help would be appreciated
    thanks
    waqas
     
  2. danardf

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    Hi.

    You have a trunk SIP with not register information.
    If you use your server behind a router, you must put the nat configuration into your trunk and Asterisk.

    into sip_nat.conf:
    externip=your ip public. or externhost=your.domain.ddns.org
    localnet=192.168.1.0/255.255.255.0

    Into your trunk conf:
    nat=yes

    Maybe that your trunk need the Register String parameter like that:

     
  3. waqasbhutta

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    its all there
     
  4. danardf

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    Strange!

    And you did make routed port from (5060 , 10000 to 20000 UDP) to @IP server?
     
  5. waqasbhutta

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    TCP: 3478,5004-5082,5222,8891
    UDP:3478,4569,10001-20000,5004-5082,5222

    they all are open for voip
    i am using this for the last 4 yrs
    never had this issue
    i thought it could be problem with my network card but then i can access my elastix from my home network

    i think its not connecting to net, but then when i do yum update , it gets updated
    thanks
     
  6. danardf

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    With these parameters, you must be able to connect on your trunk sip.
    Your problem must be on your operator SIP or your trunk parameters.

    use >sip debug peer operator SIP

    and look at the result.

    Else, are you sure that your operator need to register string?

    what's the result from :
    sip show peers? (only your operator line)
     
  7. waqasbhutta

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    another strange thing
    when i put
    sip show peers
    its not recognizing the command

    -bash: sip: command not found
     
  8. danardf

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    It's true, but before, you must go to asterisk console :p :
    # asterisk -rvvvvvvvvv
    CLI>sip show peers
    CLI>exit (to exiting)
     
  9. danardf

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    You did change my Karma? :blink: from 34 i have 33 :blush:
     
  10. waqasbhutta

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    waqas*CLI> sip show peers
    Name/username Host Dyn Nat ACL Port Status
    voicenetwork-in 64.34.135.88 N 5060 UNREACHABLE
    PoivyWaqas/badnam420 194.120.0.198 N 5060 Unmonitored
    Gizmo/waqasbhutta 198.65.166.131 N 5060 Unmonitored
    2016 (Unspecified) D N 0 UNKNOWN
    2014 (Unspecified) D N 0 UNKNOWN
    2013 (Unspecified) D N 0 UNKNOWN
    2010 (Unspecified) D N 0 UNKNOWN
    2003 (Unspecified) D N 0 UNKNOWN
    2002 (Unspecified) D N 0 UNKNOWN
    2001 (Unspecified) D N 0 UNKNOWN
    1137577235/1137577235 64.34.135.88 N 5060 UNREACHABLE
    11 sip peers [Monitored: 0 online, 9 offline Unmonitored: 2 online, 0 offline]
     
  11. danardf

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    hmmmm. On each extension you have put :nat=yes.
    Put nat=no into your LAN.
    Add qualify=yes on each extensions and into your trunk.

    For example:

    trunk name : voicenetwork
    peer detail
    live blank all other parameters (incoming setting ..Etc)

    define defaultexpirey=1800 or 3600 into sip_general_custom.conf.
     
  12. DjBac

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    It is driving me crazy!!!

    I have a similar problem. I've been for hours on the phone with my VoiP provider with no result.

    I have 3 numbers, when I have only one trunk enabled it registers just fine, when I enable 2 or more none will register.

    Why that?

    Moreover, even with one trunk I cannot get outbound calls to work.
     
  13. danardf

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    hmmm. Maybe you can try to put insecure=very into your config trunk.
    But I think that will not change to your problem.

    If only you could try another SIP operator!? :blush:

    But if you don't have OK on each status extension... (It's dead! As we say in France)
     
  14. DjBac

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    They've been very helpfull, but they are only familiar with asterisk and not freepbx, so they tell me that this might mess things up!

    It's weird though, If I have on x-lite all 3 sip account they register fine, but with asterisk nothing! If I have only one sip trunk enabled the it is fine (though with type=peer I get no inbound, only with type=user works and I cannot get any outbound calls :p )
     
  15. danardf

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    Type:
    Code:
    peer   = <--
    user   = -->
    friend = <-->
    
    For any trunk, you can put peer or friend, but not user. (usually) ;)
     
  16. DjBac

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    If I put either peer or friend when I call my voip number then I get a busy tone, only with "user" works!! :(

    Any idea on why when I have more than one sip trunks for registry they won't register?
     
  17. danardf

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    your operator need to register information?
    Any operator do not need this information!
    VoipDiscount for Example.
    This operator is free (only for one minutes by call). It's good for any test
    Just create an account, download the softphone and get the config parameters.
    Like that you will be sure that your problem come from you or from your operator.

    You can try config trunk (see the post before).

    Your operator is strange to use the user information!
    Because if you are type user, your operator must be peer. It's not common. :huh:

    Could you give me your config trunk?

    Else, another question.
    You did put the DNS information into your config network?
    Your router parameters are right?

    For try to debug this, you can enabled the debug mode like that:
    Go to CLI mode
    CLI> sip debug peer your-trunk

    And look at the result.
     
  18. danardf

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    Just for example:

    [​IMG]
     
  19. DjBac

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    I don't know what to do to help you! You made my day, really! I was so frustrated yesterday!! :p

    It was as simple as that. Now all numbers register ULTRA FAST!!! :p

    On my first install of Elastix I've changed the DNS, but on this no!!

    You are GREAT!!
     
  20. DjBac

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    My only concern no is that I cannot make outbound calls from these trunks (from mISDN is just fine). I get "all circuits are busy now". ANy more ideas!! :D
     

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