Asterisk AMP & Incoming calls

Discussion in 'UK VOIP' started by Peter Watson, May 25, 2005.

  1. Peter Watson

    Peter Watson Guest

    I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
    Speedtouch router. As a result the * box is 'visible' on a proper WAN
    IP. My IP phones are on a 10.0.0.x network, connected to eth0.

    I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and can
    make outgoing calls via either route.

    I'm using AMP to set all this up, but I can't get incoming calls (via
    Sipgate) to work. It probably doesn't help that the AMP 'Incoming
    calls' screen has greyed out radio buttons for selecting where incoming
    calls should be routed to and that the 'active' selection is 'Ring
    Group' but I've put my IP phone into Ring Group #1 and selected it. If
    I dial 7777 (AMP configured demo extension that behaves as an incoming
    call) the IP phone rings correctly but if I ring my Sipgate PSTN number
    the call doesn't go through.

    Has anyone else got this working or do I need to go back to editing the
    ..conf files by hand :)

    I've posted a debug file at http://www.pwatson.org/asterisk_log.txt if
    anyone has got time to wade through it and shed some light!

    TIA

    Peter
     
    Peter Watson, May 25, 2005
    #1
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  2. Peter Watson

    Ian Guest

    "Peter Watson" <> wrote in message
    news:4294a89d$0$39077$...
    > I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
    > Speedtouch router. As a result the * box is 'visible' on a proper WAN
    > IP. My IP phones are on a 10.0.0.x network, connected to eth0.
    >
    > I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and can
    > make outgoing calls via either route.
    >
    > I'm using AMP to set all this up, but I can't get incoming calls (via
    > Sipgate) to work. It probably doesn't help that the AMP 'Incoming
    > calls' screen has greyed out radio buttons for selecting where incoming
    > calls should be routed to and that the 'active' selection is 'Ring
    > Group' but I've put my IP phone into Ring Group #1 and selected it. If
    > I dial 7777 (AMP configured demo extension that behaves as an incoming
    > call) the IP phone rings correctly but if I ring my Sipgate PSTN number
    > the call doesn't go through.
    >
    > Has anyone else got this working or do I need to go back to editing the
    > .conf files by hand :)
    >
    > I've posted a debug file at http://www.pwatson.org/asterisk_log.txt if
    > anyone has got time to wade through it and shed some light!
    >
    > TIA
    >
    > Peter

    Hi
    relevent bits of sip.conf and extensions.conf would be a bit more usefull
    though.
    personally I only edit the conf files. Seeing no need to use a gui

    Ian
     
    Ian, May 25, 2005
    #2
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  3. Peter Watson

    Ian Guest

    "Peter Watson" <> wrote in message
    news:4294a89d$0$39077$...
    > I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
    > Speedtouch router. As a result the * box is 'visible' on a proper WAN
    > IP. My IP phones are on a 10.0.0.x network, connected to eth0.
    >

    Why didnt you just forward the relevent ports?
    Its lot easier and tried and tested.


    <snip>

    IAn
     
    Ian, May 25, 2005
    #3
  4. Peter Watson

    Peter Watson Guest

    Ian wrote:

    >
    > Why didnt you just forward the relevent ports?
    > Its lot easier and tried and tested.
    >

    Ah, that's a long story - I've just moved to Bulldog and my IP address
    is now dynamic (it changes every time my ADSL connects). I also have a
    dyndns domain so I'm running an update client to keep my IP address
    updated in their DNS. Running pptp seemed like the easiest way of
    making sure the update client gets the correct address :) I may go back
    to a conventional setup and put the Linux box in the DMZ instead....

    Peter
     
    Peter Watson, May 25, 2005
    #4
  5. Peter Watson wrote:
    || I've setup Asterisk on a Linux box that is running pptp (via eth0)
    || to my Speedtouch router. As a result the * box is 'visible' on a
    || proper WAN IP. My IP phones are on a 10.0.0.x network, connected to
    || eth0.
    ||
    || I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and
    || can make outgoing calls via either route.
    ||
    || I'm using AMP to set all this up, but I can't get incoming calls (via
    || Sipgate) to work. It probably doesn't help that the AMP 'Incoming
    || calls' screen has greyed out radio buttons for selecting where
    || incoming calls should be routed to and that the 'active' selection
    || is 'Ring Group' but I've put my IP phone into Ring Group #1 and
    || selected it. If I dial 7777 (AMP configured demo extension that
    || behaves as an incoming call) the IP phone rings correctly but if I
    || ring my Sipgate PSTN number the call doesn't go through.
    ||
    || Has anyone else got this working or do I need to go back to editing
    || the .conf files by hand :)
    ||
    || I've posted a debug file at http://www.pwatson.org/asterisk_log.txt
    || if anyone has got time to wade through it and shed some light!
    ||
    || TIA
    ||
    || Peter

    Yep.

    Working here.

    You've got to set the Sipgate number as a DID and point the DID to an
    extension or ring group.

    make sure your register for the Sipgate trunk is like this -
    register=7DigitSIPNumber:/7DigitSipNumber.

    and that your user details *&* peer details are like this-

    authuser=7DigitSipNumber
    context=ext-did
    dtmfmode=info
    fromdomain=sipgate.co.uk
    fromuser=7DigitSipNumber
    host=sipgate.co.uk
    insecure=very
    qualify=yes
    secret=8CHARACTERPASSWORD
    type=peer
    username=7DigitSipNumber

    context=ext-did

    When you create your DID, make sure it is your 7DigitSipNumber & not your
    geographic number that you use.

    There are other ways & means to do what you want, I can imagine, however,
    the above works fine here.

    ****************

    The thing I need help on is installing Perl so that I can use Webmin to set
    up Sendmail (actually I just want to set up Sendmail)
     
    The Cable Guy, May 25, 2005
    #5
  6. The Cable Guy wrote:
    || Peter Watson wrote:
    |||| I've setup Asterisk on a Linux box that is running pptp (via eth0)
    |||| to my Speedtouch router. As a result the * box is 'visible' on a
    |||| proper WAN IP. My IP phones are on a 10.0.0.x network, connected
    |||| to eth0.
    ||||
    |||| I've setup two trunks - Call1899 (via iax) and Sipgate (via sip)
    |||| and can make outgoing calls via either route.
    ||||
    |||| I'm using AMP to set all this up, but I can't get incoming calls
    |||| (via Sipgate) to work. It probably doesn't help that the AMP
    |||| 'Incoming calls' screen has greyed out radio buttons for selecting
    |||| where
    |||| incoming calls should be routed to and that the 'active' selection
    |||| is 'Ring Group' but I've put my IP phone into Ring Group #1 and
    |||| selected it. If I dial 7777 (AMP configured demo extension that
    |||| behaves as an incoming call) the IP phone rings correctly but if I
    |||| ring my Sipgate PSTN number the call doesn't go through.
    ||||
    |||| Has anyone else got this working or do I need to go back to editing
    |||| the .conf files by hand :)
    ||||
    |||| I've posted a debug file at http://www.pwatson.org/asterisk_log.txt
    |||| if anyone has got time to wade through it and shed some light!
    ||||
    |||| TIA
    ||||
    |||| Peter
    ||
    || Yep.
    ||
    || Working here.
    ||
    || You've got to set the Sipgate number as a DID and point the DID to an
    || extension or ring group.
    ||
    || make sure your register for the Sipgate trunk is like this -
    ||
    register=7DigitSIPNumber:/7DigitSipNumber.
    ||
    || and that your user details *&* peer details are like this-
    ||
    || authuser=7DigitSipNumber
    || context=ext-did
    || dtmfmode=info
    || fromdomain=sipgate.co.uk
    || fromuser=7DigitSipNumber
    || host=sipgate.co.uk
    || insecure=very
    || qualify=yes
    || secret=8CHARACTERPASSWORD
    || type=peer
    || username=7DigitSipNumber
    ||
    || context=ext-did
    ||
    || When you create your DID, make sure it is your 7DigitSipNumber & not
    || your geographic number that you use.
    ||
    || There are other ways & means to do what you want, I can imagine,
    || however, the above works fine here.
    ||
    || ****************
    ||
    || The thing I need help on is installing Perl so that I can use Webmin
    || to set up Sendmail (actually I just want to set up Sendmail)

    Bad form to reply to my own post, I know.

    I assumed you would clearly understand to replace 7DigitSipNumber with your
    actual 7 digit sip number etc.....
     
    The Cable Guy, May 25, 2005
    #6
  7. Peter Watson

    Roly Guest

    "Peter Watson" <> wrote in message
    news:4294a89d$0$39077$...
    > I've setup Asterisk on a Linux box that is running pptp (via eth0) to my
    > Speedtouch router. As a result the * box is 'visible' on a proper WAN IP.
    > My IP phones are on a 10.0.0.x network, connected to eth0.
    >
    > I've setup two trunks - Call1899 (via iax) and Sipgate (via sip) and can
    > make outgoing calls via either route.
    >
    > I'm using AMP to set all this up, but I can't get incoming calls (via
    > Sipgate) to work. It probably doesn't help that the AMP 'Incoming calls'
    > screen has greyed out radio buttons for selecting where incoming calls
    > should be routed to and that the 'active' selection is 'Ring Group' but
    > I've put my IP phone into Ring Group #1 and selected it. If I dial 7777
    > (AMP configured demo extension that behaves as an incoming call) the IP
    > phone rings correctly but if I ring my Sipgate PSTN number the call
    > doesn't go through.
    >
    > Has anyone else got this working or do I need to go back to editing the
    > .conf files by hand :)
    >
    > I've posted a debug file at http://www.pwatson.org/asterisk_log.txt if
    > anyone has got time to wade through it and shed some light!
    >
    > TIA
    >
    > Peter


    Here are the two screenshots from my setup. The bit that miffed me for a
    while was that seemingly a DID route needs to be defined.

    http://makeashorterlink.com/?D1C71242B
    and
    http://makeashorterlink.com/?L2E71542B

    Roly.
     
    Roly, May 25, 2005
    #7
  8. Peter Watson

    Peter Watson Guest

    Ian wrote:

    >
    > Hi
    > relevent bits of sip.conf and extensions.conf would be a bit more usefull
    > though.
    > personally I only edit the conf files. Seeing no need to use a gui
    >

    I've now uploaded the .conf files as http://www.pwatson.org/conf_files.zip

    Thanks,

    Peter
     
    Peter Watson, May 25, 2005
    #8
  9. Peter Watson

    Ian Guest

    "Peter Watson" <> wrote in message
    news:4294b834$0$557$...
    > Ian wrote:
    >
    > >
    > > Hi
    > > relevent bits of sip.conf and extensions.conf would be a bit more

    usefull
    > > though.
    > > personally I only edit the conf files. Seeing no need to use a gui
    > >

    > I've now uploaded the .conf files as http://www.pwatson.org/conf_files.zip
    >
    > Thanks,
    >

    Only had time to skim them, But cant see any mention of incoming sipgate
    number in the extensions.conf. it need to be in there it wont be handled by
    the s extension

    Ian
     
    Ian, May 25, 2005
    #9
  10. Peter Watson

    Peter Watson Guest

    In article <4294b7eb$0$580$>,
    says...

    >
    > Here are the two screenshots from my setup. The bit that miffed me for a
    > while was that seemingly a DID route needs to be defined.
    >
    > http://makeashorterlink.com/?D1C71242B
    > and
    > http://makeashorterlink.com/?L2E71542B
    >
    > Roly.
    >
    >

    Thanks for all the relplies to this thread - I've altered my outgoing
    registration stuff and now incoming calls work!

    Peter
     
    Peter Watson, May 26, 2005
    #10
  11. Peter Watson

    Peter Watson Guest

    In article <>,
    says...

    > Thanks for all the relplies to this thread - I've altered my outgoing
    > registration stuff and now incoming calls work!
    >
    > Peter
    >

    Well at least they did until I changed my network config back to my *
    box being in the DMZ of my router instead of running PPTP...

    <Much gnashing of teeth!>

    Peter
     
    Peter Watson, May 29, 2005
    #11
  12. Peter Watson

    Peter Watson Guest

    In article <>,
    says...
    > In article <>,
    > says...
    >
    > > Thanks for all the relplies to this thread - I've altered my outgoing
    > > registration stuff and now incoming calls work!
    > >
    > > Peter
    > >

    > Well at least they did until I changed my network config back to my *
    > box being in the DMZ of my router instead of running PPTP...
    >
    > <Much gnashing of teeth!>
    >
    > Peter
    >

    Scrub that - the SIP 'handler' in my Speedtouch 510 had turned itself
    back on and prevented * registering with Sipgate

    <Even more gnashing of teeth!!>
     
    Peter Watson, May 29, 2005
    #12
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