Unable to call on SIP Trunks

Discussion in 'General' started by jessie, Sep 17, 2008.

  1. jessie

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    Please help, I've trying to send a call on SIP trunks but I always get "all circuits are busy". Could anyone have a working SIP trunk config and extensions config?
     
  2. busster8

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    Unfortunately, each sip trunk provider is different. There is no universal sip trunk setup. Who is the provider?
     
  3. jessie

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    I have two providers, one gafachi and telopar. they send me the parameters and required to input on the sip.conf, the aithentication method is two, 1.IP authentication only and 2.with username and password. But in elastix, sip.conf is not allowed to have hand edited, even on the extensions.conf. For this, how am I going to set the parameters if GUI config inputs is not working?
     
  4. busster8

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    What did the providers send you?

    X out the confidential information.
     
  5. jessie

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    this are the option:

    Configuration Examples for Asterisk

    Listed below are different examples based on version of Asterisk and Caller ID options.
    Any version of Asterisk - No Caller ID
    Asterisk version below 1.2 - sending a specific Caller ID
    Asterisk version below 1.2 - Caller ID pass-thru
    Asterisk version 1.2 and above - sending a specific Caller ID
    Asterisk version 1.2 and above - Caller ID pass-thru


    Hence, Elastix used Asterisk above version, I select eh Asterisk version 1.2 and above -sending a specific Caller ID. The parameter are below:

    Configuration Example for an Asterisk version 1.2 and above - sending a specific Caller ID
    (Replace the 2125551212 with the 10-digit number you wish to send)

    /etc/asterisk/extensions.conf
    [general]
    static=yes
    writeprotect=yes

    [default]
    exten => _1NXXNXXXXXX,1,Set(CALLERID(number)=2125551212)
    ; NOTE: The line below is required. It will not affect the Calling Name
    ; from displayed to the called party. If the called party subscribes to
    ; Calling Name service, the called party's local exchange carrier performs
    ; a query to determine the Calling Name based on the Calling Number that is
    ; received with the call.
    exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=)
    exten => _1NXXNXXXXXX,3,Dial,SIP/${EXTEN}@gafachi
    exten => _1NXXNXXXXXX,4,Hangup
    exten => _011.,1,Set(CALLERID(number)=2125551212)
    exten => _011.,2,Set(CALLERID(name)=)
    exten => _011.,3,Dial,SIP/${EXTEN}@gafachi
    exten => _011.,4,Hangup

    [gafachi-incoming]
    ; NOTE: this section is for Gafachi Origination (incoming) services.
    ; DID and/or Toll free origination numbers need to include the "1" prefix
    ; Replace the 12125551234 with your DID or Toll Free number, and repeat or
    ; change as needed.
    exten => 12125551234,1,Dial,Zap/g1
    exten => 12125551234,2,Hangup



    /etc/asterisk/sip.conf
    [general]
    port=5060
    bindaddr=0.0.0.0
    context=default
    tos=lowdelay
    disallow=all
    allow=ulaw

    register=>xxxxxxxxx:Xxxxxxxxxx@sip.gafachi.com
    ; NOTE: The line below ([gafachi]) can not be changed, otherwise your Asterisk
    ;system will reject calls, with a "403 Forbidden", from the Gafachi Network.
    [gafachi]
    type=friend
    username=xxxxxxxxxxxx
    secret=Xxxxxxxxxx
    host=sip.gafachi.com
    canreinvite=no
    fromuser=xxxxxxxxx
    dtmfmode=rfc2833
    context=gafachi-incoming

    Here is the another issue, on the asterisk used in Elastix, it is indicated that:

    ; do not edit this file, this is an auto-generated file by freepbx
    ; all modifications must be done from the web gui
    ;

    If it is not recommended to hand edit sip.conf file, how am i going to set the paramters of the VoIP Provider? On the GIU freepbx interface, I already add up the trunk SIP with the same parameters given by Gafachi.

    This is the same scenario in extensions.conf, I dont know where to put the dial plan.

    Your time is much appreciated.
     
  6. hendrikdw

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    Hi Guys

    I have a different problem.
    Currently I have 2 outgoing trunks.
    1 - ZAP PRI
    2 - SIP Trunk
    But here is my problem. If I send a fax, I want to force it to use the ZAP trunk and not the SIP trunk.
    How will I do this?

    Thanks
     
  7. Chilling_Silence

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    hendrikdw mate,

    You should have started a new thread, new problem!

    Anyway, easiest way Ive found is to prefix all Faxes with 9, and have a dedicated outbound trunk that you can use that just has 9|.
    That's mostly because Ive setup my dial-plan so it dials all numbers without requiring a user to dial 1 or 9 to get an outside number ;)

    You might want to try it with 8|. for faxes instead in your outbound routes?
     

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