Can not make outbound calls at all, help!?

Discussion in 'General' started by Darkness, Jan 9, 2009.

  1. Darkness

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    Please I don't know all these technical terms and stuff. I just want to know how I can fix my outbound calls problem as quickly as possible. I have set up my trunks and the incomming and outgoing routes and I don't know what I did wrong but I can't seem to call out we can only receive incomming calls.

    Help?!
    P.S. Couldn't you make elastix easier?:blush:
     
  2. abanuz

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    hi;

    Elastix support different trunk types.What is your trunk type?..You want to use sip trunk.?


    Firstly;

    You create a trunk for sip.And check account,its register or not.(you can check via cli...You can login bye ssh and type asterisk -r and than "sip show registery")...if its look like register,you just create outbound route and select this trunk under the web page.(Like "9|." according this,if you press 9 and dial number,call will go the trunk)..

    Thanks
     
  3. Bob

    Bob

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    Darkness,

    We have to assume that you are using SIP connections to a Voice Provider. The fact that you are able to receive incoming calls (from the fact that you only mention outgoing calls is the issue), means that you have most of the info in the SIP trunks as the Voice Provider knows what IP address your PBX is and is able to forward calls.

    It actually sounds like your SIP trunk is not registered, as many providers will still provide calls to you, but will not let you dial out unless you are registered). The other possibility is that you have turned on Allow Anonymous SIP (on General page), which means that your trunk registration is actually not properly setup, and it is only by minimal configuration that your calls are coming in.

    If you haven't already, have a look at Elastix without tears (available in the Elastix downloads area), and have a very good read. I don't mean a skim over. If you don't understand it, then you are going to continually run into areas that will be too complex.

    Elastix itself is actually very easy to setup, it is the number of options/connections/hardware which Elastix (or any other distribution) cannot account for and can never be expected to. Whilst Elastix is relatively simple to setup using an ISO, PBX functionality, in particular VoIP is never going to be a simple task, unless you have the internet connection, router, network, mail server, software, hardware, voice provider, configuration all provided from one company. In reality this is basically not possible, which is why I am pointing out that you have to have some understanding.

    So going back to your question, it appears that your trunk is probably your main issue (unless you have no outbound routes). I would start by asking about the trunk, providing examples of your trunk (without passwords), and asking others for help.

    Regards

    Bob
    P.S Couldn't you make it easier for others to help you with a little more information at the start instead of making us guess...:)
     
  4. snooth

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    Hi Bob,

    I have the same issue as Darkness and this is a fresh install of elastix 2.2.0.

    Here are my Pennytel Trunk settings

    username=XXXXXXXXX
    type=peer
    secret=XXXXXX
    insecure=very
    host=sip.pennytel.com
    disallow=all
    allow=alaw&ulaw
    canredirect=no
    canreinvite=no


    I can receive incoming calls but no outgoing calls, it keep saying all circuits are busy now...

    thanks in advanced to any help.
     
  5. jgutierrez

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    Did you set the outbound callerID on the SIP trunk?
     
  6. rgranados

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    Re: Re:Can not make outbound calls at all, help!?

    Snooth:

    In your configuration i can see that you are using g711 codec, there are some SIP providers that do not manage that codec due to the bandwidth consumption, most accepted codec is g729. Have you checked with your SIP Trunk provider that is accepting g711?

    Best Regards...
     
  7. Bob

    Bob

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    Snooth,

    Can you give these a try....its been while since I have used Pennytel (naturally keep a backup of what you have......)....

    Let me know the results as just looking at it, it may need some further lines, but try it first as I was using it a couple of years ago and it was working...


    Peer Details [PennyTel]

    type=peer
    disallow=all
    allow=alaw&ulaw&g729
    qualify=no
    canredirect=no
    canreinvite=no
    host=sip.pennytel.com
    insecure=very
    secret={password}
    username={username}
    dtmfmode=rfc2833


    User Details [Username]
    type=user
    context=from-trunk
    secret={password}

    Register String
    {username}:{password}@sip.pennytel.com/{username}


    Regards

    Bob
     
  8. snooth

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    Hi Bob,

    Nope, that didn't make a difference.. still the same issue.

    Thanks.
     
  9. Bob

    Bob

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    Snooth,

    Under the Asterisk CLI.....

    type

    sip show peers

    &

    sip show registry

    and post the results

    Regards

    Bob
     
  10. snooth

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    Bob,

    sip show peers

    Name/username Host Dyn Forcerport ACL Port Status
    20/20 (Unspecified) D N A 0 UNKNOWN
    21/21 10.1.1.4 D N A 5060 OK (5 ms)
    22/22 10.1.1.5 D N A 5060 OK (5 ms)
    61XXXXXXXXX /61XXXXXXXXX 202.85.243.105 N 5060 Unmonitored
    Pennytel /61XXXXXXXXX 202.85.243.105 N 5060 Unmonitored
    5 sip peers [Monitored: 2 online, 1 offline Unmonitored: 2 online, 0 offline]


    sip show registry

    Host dnsmgr Username Refresh State Reg.Time
    sip.pennytel.com5060:5060 N 61XXXXXXXXXX 120 Request Sent
    1 SIP registrations.


    Here are the outputs, all looks good, just not sure why it doesn't connect on outgoing calls.. incoming is fine!

    the last think I can post is a output of a failed call.

    Thanks for your time Bob.
     
  11. Bob

    Bob

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    Snooth,

    Basically its saying that it has not registered.

    Basically it has sent a request to register, but for some reason it has not seen a reply back.

    Off the top of my head, possibly a router configuration issue or your registration details are incorrect...

    Have a look at the fault finding doc in my signature and turn on sip debugging for the trunk to work out what is happening to the registration. Need to confirm that the packets are coming back to the Elastix system, and what the packets are saying the error is...

    Regards

    Bob
     
  12. snooth

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    Hi Bob,

    that was my bad, I made some changes to see if it worked by I had a typo.. look in the register string, i didn't put a : between the .com5060, should be .com:5060, after changing that, it has registered.

    Here is the strange thing, Pennytel use to work with an older version of Elastix, both incoming and outgoing calls, now with this newer version of elastix 2.2, same configurations etc.. no outgoing calls with pennytel only incoming calls.

    However if i use a different provider, i.e. Gotalk, then both incoming and outgoing calls work on elastix 2.2.

    I wonder what has changed since elastix 2.2 to make pennytel stop making outgoing calls??

    Any ideas? otherwise i'm going to have to sniff packets... I long process I don't want to have to do..

    Thanks guys!
     
  13. Bob

    Bob

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    Snooth,

    OK you are now registered.....

    Can you make a call out and post the relevant part of the log /var/log/asterisk/full so we can see what is happening.

    A reasonable number of differences between Asterisk 1.6 and 1.8...

    But sometimes the hat is hung on that, when it is possible that other things come into play, such as small mistakes (not saying that is your issue)....just a case of debugging....

    So lets go with the log first, and then look at SIP Debug if we don't find anything...

    Regards

    Bob
     

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