Trixbox SQL database

Discussion in 'UK VOIP' started by Alister, May 21, 2007.

  1. Alister

    Alister Guest

    I have been tasked with setting up a VOIP solution for our office
    comms.

    As a relative newbie to voip, and because of previous attempts within
    the company, we are using a Trixbox
    installation on one of our servers. Both asterisk and trixbox are the
    latest versions.

    my problem is that, using the FreePBX web front end to setup
    extensions, the entries in various .conf files don't work with the
    telephones we have (atcom at320) and I have to go into config edit and
    manually re-write the sip_additional.conf and
    extensions_additional.conf to get them to work. This however doesn't
    update the MySQL database that trixbox uses, so everytime I make any
    changes via freepbx I have to copy and paste my copy of the conf files
    back into asterisk.

    The symptoms are that on dialling a number - internal or external, the
    phones go direct to busy signal.
    Using a softphone, the log shows "Call rejected".

    The bits that i have to manually edit are to do with my outgoing route
    contexts in extensions_additional.conf
    and the allowed codecs in sip_additional.conf.

    As an example - the outgoing route extensions don't work as freepbx
    writes them:

    [outrt-001-PSTN Out]
    include => outrt-001-PSTN Out-custom
    exten => _0XXXXXXXXXX,1,Macro(dialout-trunk,2,${EXTEN},,)
    exten => _0XXXXXXXXXX,n,Macro(outisbusy,)

    ; end of [outrt-001-PSTN Out]

    I have to change it to:

    [outrt-001-PSTN Out]
    include => outrt-001-PSTN Out-custom
    exten => _0XXXXXXXXXX,1,Dial(SIP/voiptalk-out/${EXTEN})
    exten => _0XXXXXXXXXX,n,Macro(outisbusy,)

    ; end of [outrt-001-PSTN Out]

    I am sure I am doing something wrong - I just need you to tell me what
    (and probably what an idiot I am).

    Thanks

    Alister
    Alister, May 21, 2007
    #1
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  2. Alister

    Desk Rabbit Guest

    Alister wrote:
    > I have been tasked with setting up a VOIP solution for our office
    > comms.
    >
    > As a relative newbie to voip, and because of previous attempts within
    > the company, we are using a Trixbox
    > installation on one of our servers. Both asterisk and trixbox are the
    > latest versions.
    >
    > my problem is that, using the FreePBX web front end to setup
    > extensions, the entries in various .conf files don't work with the
    > telephones we have (atcom at320) and I have to go into config edit and
    > manually re-write the sip_additional.conf and
    > extensions_additional.conf to get them to work. This however doesn't
    > update the MySQL database that trixbox uses, so everytime I make any
    > changes via freepbx I have to copy and paste my copy of the conf files
    > back into asterisk.


    I suspect the clue is in the first line of the config files:-

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

    The changes are made in the GUI which writes to the SQL Database. When
    you apply the changes, it reads from the SQL database and writes the
    config files which asterisk uses.

    > As an example - the outgoing route extensions don't work as freepbx
    > writes them:
    >
    > [outrt-001-PSTN Out]
    > include => outrt-001-PSTN Out-custom
    > exten => _0XXXXXXXXXX,1,Macro(dialout-trunk,2,${EXTEN},,)
    > exten => _0XXXXXXXXXX,n,Macro(outisbusy,)
    >
    > ; end of [outrt-001-PSTN Out]
    >
    > I have to change it to:
    >
    > [outrt-001-PSTN Out]
    > include => outrt-001-PSTN Out-custom
    > exten => _0XXXXXXXXXX,1,Dial(SIP/voiptalk-out/${EXTEN})
    > exten => _0XXXXXXXXXX,n,Macro(outisbusy,)
    >
    > ; end of [outrt-001-PSTN Out]


    I'd say that's more likely a problem with the way you have defined your
    trunks and/or routes as the way Freepbx writes the config files is not
    an issue.
    Desk Rabbit, May 21, 2007
    #2
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  3. Alister

    alexd Guest

    Alister wrote:

    > I have been tasked with setting up a VOIP solution for our office
    > comms.
    >
    > As a relative newbie to voip, and because of previous attempts within
    > the company, we are using a Trixbox
    > installation on one of our servers. Both asterisk and trixbox are the
    > latest versions.
    >
    > my problem is that, using the FreePBX web front end to setup
    > extensions, the entries in various .conf files don't work with the
    > telephones we have (atcom at320)


    I've got one at home which works fine with vanilla Asterisk. I also have two
    at work on a Trixbox 2 box. At neither site do they require any kind of
    special fiddling to get them to work.

    > and I have to go into config edit and
    > manually re-write the sip_additional.conf and
    > extensions_additional.conf to get them to work. This however doesn't
    > update the MySQL database that trixbox uses, so everytime I make any
    > changes via freepbx I have to copy and paste my copy of the conf files
    > back into asterisk.
    >
    > The symptoms are that on dialling a number - internal or external, the
    > phones go direct to busy signal.
    > Using a softphone, the log shows "Call rejected".


    Do you have any extensions that aren't AT-320's? Perhaps you could install
    X-Lite or similar on a PC to test it out? Have you put them on the latest
    firmware?

    > I am sure I am doing something wrong - I just need you to tell me what
    > (and probably what an idiot I am).


    Try and forget about the text config files. If all you're trying to do is
    get two extensions to work with an IP trunk, you shouldn't need to resort
    to making manual changes to the .conf files. Try and get it working through
    the web interface.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    15:59:17 up 22 days, 17:59, 2 users, load average: 1.65, 1.10, 0.62
    09 f9 11 02 9d 74 e3 5b d8 41 56 c5 63 56 88 c0
    alexd, May 21, 2007
    #3
  4. Alister

    Alister Guest

    On 21 May, 09:31, Alister <> wrote:
    <snip>

    Well, I did say I was an idiot!

    Thanks to Desk Rabbit and alexd for your replies.

    I have now found the rather silly errors I was making, and all works
    fine.

    In sip.conf, in the [general] section I had put disallow=all - trying
    to get an awkward mobile phone to work so
    I was changing the allowed codecs on a phone by phone basis. I forgot
    to allow the various codecs again
    globally when I had finished - doh!

    In extensions.conf I had taken out the include=my_outgoing_context for
    some daft reason.

    All in all not a good day.

    Just as a matter of interest though, is there any way of updating the
    SQL database from manually altered
    conf files?

    Thanks

    Alister
    Alister, May 21, 2007
    #4
  5. Alister

    alexd Guest

    Alister wrote:

    > Just as a matter of interest though, is there any way of updating the
    > SQL database from manually altered conf files?


    No.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    21:04:05 up 22 days, 23:04, 2 users, load average: 0.58, 0.54, 0.54
    09 f9 11 02 9d 74 e3 5b d8 41 56 c5 63 56 88 c0
    alexd, May 21, 2007
    #5
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