Troubleshooting - how to start

Discussion in 'General' started by justconnect, Jun 17, 2009.

  1. justconnect

    Joined:
    Jun 17, 2009
    Messages:
    11
    Likes Received:
    0
    Hope one of the "old hands" will have an answer to this!

    The problem I am currently battling with is that I am unable to make outbound calls on a SIP trunk. My Voip Service Provider is unfortunately not able to provide me with the details I require to enter for peer and user details when setting up the SIP trunk. My questions are:

    1) Is there any way to work out what these settings ought to be? If not, what settings are LIKELY to work?
    2) How would one go about trouble shooting this issue? Perhaps if I could create a log and dig through it I could figure out where the problem lies.

    Just to add - I have successfully managed to dial using my sip username & password using a softphone so the SIP account is live and working.

    Many thanks for any assistance!
     
  2. Patrick_elx

    Joined:
    Dec 14, 2008
    Messages:
    1,120
    Likes Received:
    0
    what did you set up in your trunk?


    You can look at the log at: /var/log/asterisk/full

    or better just log on your server
    type asterisk -rvvv to go in the cli
    type sip set debug ip IPADDRESS where IPADDRESS is the ipaddress or name of your provider gateway
    and look at what's happening
     
  3. rafael

    Joined:
    May 14, 2007
    Messages:
    1,454
    Likes Received:
    1
    The commando
    tail -f /var/log/asterisk/full

    Would show you the logs in real time.
     
  4. justconnect

    Joined:
    Jun 17, 2009
    Messages:
    11
    Likes Received:
    0
    Hi there

    Ok - when I go to Cli and type "sip debug ip 196.30.127.180". I then get a message saying "SIP Debugging Enabled for IP: 196.30.127.180"

    If I wish to actually SEE the log, how do I access this? I cannot seem to find a place within the GUI to see these logs!

    Thanks
     
  5. Patrick_elx

    Joined:
    Dec 14, 2008
    Messages:
    1,120
    Likes Received:
    0
    You have activated the sip log, now you just have to wait for some activity on 196.30.124.180.
    If nothing is happening, you probably haven't setup your trunk properly.

    I'm asking the question again:
    what did you set up for your trunk?
     
  6. justconnect

    Joined:
    Jun 17, 2009
    Messages:
    11
    Likes Received:
    0
    Hi Patrick

    Ok - here is how I setup my trunk:

    SIP TRUNK
    No dial rules
    No Outgoing Dial prefix

    Peer Details are:
    type=peer
    username=278780%%%%%
    secret=PASSWORD
    qualify=yes
    nat=yes
    host=196.30.127.180
    fromuser=278780%%%%%
    dtmfmode=rfc2833
    disallow=all
    context=from-trunk
    canreinvite=no
    allow=g729&alaw&ulaw
    insecure=very

    Incoming settings I have left blank as I am just trying to get outgoing working for now.

    Finally my register string is 278780%%%%%:pASSWORD@196.30.127.180/278780%%%%%

    On my outbound routes, I have selected the above trunk. My dial patterns are
    .
    NXXNXXXXXX
    NXXXXXX
    XXXXXXXXXX

    The rest I have left standard.

    When I try to dial a local number I get the message saying "All circuits are busy now. Please try your call again later."

    Any guidance is appreciated.

    Thanks
     
  7. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    If you want 7 digit dialing you will almost certainly have to set up your outbound trunk to add your area-code to local calls, SIP providers for the most part require explicit dialing as they don't limit there provisioning to one area code so how would they know to which area code do you want your call sent to?.

    So: your dial plan:

    NXXNXXXXXX
    NXXXXXX
    XXXXXXXXXX


    a more explicit way:

    NXXNXXXXXX
    <Your area code>+NXXXXXX

    Your last line is erroneous the first line is more explicit (and correct) anything that that doesn't match line 1 but matches line 3 is not a real phone number in the NANP (North American Numbering Plan)

    p.s.
    Line 1 also includes some "Very" expensive possibilities to the Caribbean however.
    It is often a good idea to build separate routes for "local" "long distance" "International (including Canada)" and "The Islands" so one has better control of who can dial what. (add a password for the "Islands")

    Just my 2cents worth as ever!
     
  8. jgutierrez

    Joined:
    Feb 28, 2008
    Messages:
    5,737
    Likes Received:
    0
    justconnect,
    I see on your configuration, that you have set to use g729, one quick question? Have you already installed g729? It doesn't come with Elastix, you need to download it and install it, açif you have already done that, does your provider supports it?

    I would try the following configuration on Peer Details:
    type=peer
    username=278780%%%%%
    secret=PASSWORD
    qualify=yes
    host=196.30.127.180
    fromuser=278780%%%%%
    context=from-trunk
    insecure=very
     
  9. justconnect

    Joined:
    Jun 17, 2009
    Messages:
    11
    Likes Received:
    0
    Hi jgutierrez!

    Thanks for the suggestion. Actually NO, I wasn't aware that I needed to install G729! But now obviously I am. So I have ordered 1 license to install from Digium. (I see that there are open source free ones but for $10, I figure I am not going to play around - maybe later once things are working as expected!)

    Just to update my status, I have managed to get to a point where the landline rings and I can answer but there is no audio. I assume that this is exactly the issue I would get if I didn't have the codec installed?

    Much appreciate the tip! Once the license key arrives and I have installed it, I will update my status!
     

Share This Page