VoIPDiscount with Asterisk - Success!

Discussion in 'UK VOIP' started by Sparks, Oct 25, 2005.

  1. Sparks

    Sparks Guest

    Okay, since VoIPDiscount have started charging, I have had another play

    Success!!

    My configurations is as follows...

    Asterisk@home 1.5 running on a local IP behind a router.

    The following ports are forwarded to the asterisk box
    5060 (UDP)
    8000->8010 (UDP)
    4560->4570 (UDP)
    4560->4570 (TCP)

    ------------

    In my sip.conf I have the following extra lines (Maintanace, Config Edit,
    sip.conf)

    externip=MY ROUTERS EXTERNAL IP ADDRESS or host name if you have one
    localnet=192.168.1.0/255.255.255.0 (change this to suit your network)
    canreinvite=no

    ---------

    If something is not mentioned, it is either blank, or at it's default
    setting
    If something is in (round brackets), it is for information and should be
    replaced with your details, or omitted.

    Trunks -> sipdiscount

    Maximum Channels - 1

    Dial Rules - 0044+XXXXXXXXXX
    004420+XXXXXX (Replace 20 with your area code, minus the
    leading 0)
    0044+800.
    0044+808.

    Trunk Name - sipdiscount

    PEER Details -
    allow=ulaw&alaw
    authuser=(your username)
    disallow=all
    fromdomain=sipdiscount.com
    fromuser=(your username)
    host=sip.sipdiscount.com
    insecure=very
    nat=yes
    qualify=yes
    secret=(your password)
    type=peer
    username=(your username)


    Everything else blank

    --------------------
    Outbound Routing -> sipdiscount

    Dial Patterns
    00441.
    00442.
    00443.
    0|800.
    0|808.
    0|1.
    0|2.
    0|3.
    [2-8].


    Trunk Sequence
    SIP/sipdiscount

    -----------------

    With this setup, you can just dial UK land lines & freephone numbers via
    sipdiscount like a normal phone (no prefixing required!)


    Hope this helps!
    Sparks....
     
    Sparks, Oct 25, 2005
    #1
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  2. Sparks

    Sparks Guest

    The subject should, of course, read..
    SIP Discount with Asterisk - Success!

    Doh!
     
    Sparks, Oct 25, 2005
    #2
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  3. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    Sparks wrote:
    || Okay, since VoIPBuster have started charging, I have had another
    || play
    ||
    || Success!!
    ||

    I just get "All Circuits Are Busy Now"
     
    Jono, Oct 25, 2005
    #3
  4. Sparks

    Sparks Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    >
    > Sparks wrote:
    > || Okay, since VoIPBuster have started charging, I have had another
    > || play
    > ||
    > || Success!!
    > ||
    >
    > I just get "All Circuits Are Busy Now"


    Still working OK for me now!

    One ammendment, in the Trunks -> sipdiscount

    The line "004420+XXXXXX"
    should be "004420+XXXXXXXX"

    I have also found, if you set the "Maximum Channels" to more than 1, you can
    make multtiple calls at the same time :)

    Sparks...
     
    Sparks, Oct 25, 2005
    #4
  5. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    Sparks wrote:
    ||| Sparks wrote:
    ||||| Okay, since VoIPBuster have started charging, I have had another
    ||||| play
    |||||
    ||||| Success!!
    |||||
    |||
    ||| I just get "All Circuits Are Busy Now"
    ||
    || Still working OK for me now!
    ||
    || One ammendment, in the Trunks -> sipdiscount
    ||
    || The line "004420+XXXXXX"
    || should be "004420+XXXXXXXX"
    ||
    || I have also found, if you set the "Maximum Channels" to more than 1,
    || you can make multtiple calls at the same time :)
    ||
    || Sparks...

    Encouraging.............

    I picked up on the missing X, though not living in the London area, the
    number of Xs were right for me!

    Anyway, where exactly did you put the three extra lines in your .conf file?

    Could you post your entire sip.conf file? (personal stuff deleted, of
    course)

    when you say ammend 192.168.1.0 to suit, what did you mean? Is this your
    gateway address?

    I can't even get the Test login to work!
     
    Jono, Oct 25, 2005
    #5
  6. Sparks

    Sparks Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)


    > Encouraging.............
    >
    > I picked up on the missing X, though not living in the London area, the
    > number of Xs were right for me!
    >
    > Anyway, where exactly did you put the three extra lines in your .conf
    > file?
    >
    > Could you post your entire sip.conf file? (personal stuff deleted, of
    > course)
    >
    > when you say ammend 192.168.1.0 to suit, what did you mean? Is this your
    > gateway address?


    Nope, this is your local network address.

    So for example, if all your PC's on your network were in the 10.12.25.X
    range, you would use localnet=10.12.25.0/255.255.255.0

    If you PC's were in the 10.12.X.X range you would use
    localnet=10.12.0.0/255.255.0.0

    If you are on and ADSL connection with a dynamic address, it is best to
    register with a service like www.no-ip.org
    they will (for free) provide you with a DNS name (like whatever.no-ip.org)
    then you use this in your externip= line.
    You then need to install a cluent that will update this DNS address if your
    IP address changes (otherwise you will need to change the externip= line
    every time your IP address changes) (See www.no-ip.org for sutable clients)

    > I can't even get the Test login to work!


    Okay, here is the entire sip.conf...

    -----------------
    ; Note: If your SIP devices are behind a NAT and your Asterisk
    ; server isn't, try adding "nat=1" to each peer definition to
    ; solve translation problems.

    [general]

    port = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    disallow=all
    externip=XXX.XXX.XXX.XXX ; Change this to your router's IP address, or your
    DNS name
    localnet=192.168.1.0/255.255.255.0
    allow=ulaw
    allow=alaw
    context = from-sip-external ; Send unknown SIP callers to this context
    callerid = Unknown
    canreinvite=no

    #include sip_nat.conf
    #include sip_custom.conf
    #include sip_additional.conf
    ----------------
     
    Sparks, Oct 26, 2005
    #6
  7. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    "Sparks" <postmaster@127.0.0.1> wrote in message
    news:435f4315$0$38038$...
    >
    >> Encouraging.............
    >>
    >> I picked up on the missing X, though not living in the London area, the
    >> number of Xs were right for me!
    >>
    >> Anyway, where exactly did you put the three extra lines in your .conf
    >> file?
    >>
    >> Could you post your entire sip.conf file? (personal stuff deleted, of
    >> course)
    >>
    >> when you say ammend 192.168.1.0 to suit, what did you mean? Is this your
    >> gateway address?

    >
    > Nope, this is your local network address.
    >
    > So for example, if all your PC's on your network were in the 10.12.25.X
    > range, you would use localnet=10.12.25.0/255.255.255.0
    >
    > If you PC's were in the 10.12.X.X range you would use
    > localnet=10.12.0.0/255.255.0.0
    >
    > If you are on and ADSL connection with a dynamic address, it is best to
    > register with a service like www.no-ip.org
    > they will (for free) provide you with a DNS name (like whatever.no-ip.org)
    > then you use this in your externip= line.
    > You then need to install a cluent that will update this DNS address if
    > your IP address changes (otherwise you will need to change the externip=
    > line every time your IP address changes) (See www.no-ip.org for sutable
    > clients)
    >
    >> I can't even get the Test login to work!

    >
    > Okay, here is the entire sip.conf...
    >
    > -----------------
    > ; Note: If your SIP devices are behind a NAT and your Asterisk
    > ; server isn't, try adding "nat=1" to each peer definition to
    > ; solve translation problems.
    >
    > [general]
    >
    > port = 5060 ; Port to bind to (SIP is 5060)
    > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    > disallow=all
    > externip=XXX.XXX.XXX.XXX ; Change this to your router's IP address, or
    > your DNS name
    > localnet=192.168.1.0/255.255.255.0
    > allow=ulaw
    > allow=alaw
    > context = from-sip-external ; Send unknown SIP callers to this context
    > callerid = Unknown
    > canreinvite=no
    >
    > #include sip_nat.conf
    > #include sip_custom.conf
    > #include sip_additional.conf
    > ----------------
    >
    >
    >


    Hi Sparks,

    Thanks for this, I'll check out your suggestions.

    I already have a dyndns address.
     
    Jono, Oct 26, 2005
    #7
  8. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    "Sparks" <postmaster@127.0.0.1> wrote in message news:435f4315$0
    >
    > ; Note: If your SIP devices are behind a NAT and your Asterisk
    > ; server isn't, try adding "nat=1" to each peer definition to
    > ; solve translation problems.
    >
    > [general]
    >
    > port = 5060 ; Port to bind to (SIP is 5060)
    > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    > disallow=all
    > externip=XXX.XXX.XXX.XXX ; Change this to your dyndns name
    > localnet=192.168.1.0/255.255.255.0
    > allow=ulaw
    > allow=alaw
    > context = from-sip-external ; Send unknown SIP callers to this context
    > callerid = Unknown
    > canreinvite=no
    >
    > #include sip_nat.conf
    > #include sip_custom.conf
    > #include sip_additional.conf
    > ----------------


    Hi Sparks,

    Here's my sip.conf - I can't see anything different (can you?)

    I've got all the same ports as you forwarded in the router.............but
    it still doesn't work :-(

    [general]

    port = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    disallow=all
    externip=my.dyndns.address
    localnet=192.168.1.0/255.255.255.0
    allow=ulaw
    allow=alaw
    context = from-sip-external ; Send unknown SIP callers to this context
    callerid = Unknown
    canreinvite=no

    #include sip_nat.conf
    #include sip_custom.conf
    #include sip_additional.conf
     
    Jono, Oct 26, 2005
    #8
  9. Sparks

    Sparks Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    > Hi Sparks,
    >
    > Here's my sip.conf - I can't see anything different (can you?)
    >
    > I've got all the same ports as you forwarded in the router.............but
    > it still doesn't work :-(
    >
    > [general]
    >
    > port = 5060 ; Port to bind to (SIP is 5060)
    > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    > disallow=all
    > externip=my.dyndns.address
    > localnet=192.168.1.0/255.255.255.0
    > allow=ulaw
    > allow=alaw
    > context = from-sip-external ; Send unknown SIP callers to this context
    > callerid = Unknown
    > canreinvite=no
    >
    > #include sip_nat.conf
    > #include sip_custom.conf
    > #include sip_additional.conf


    Looks fine to me!

    Have you reloaded the config files in asterisk since you made changed?
    (Easiest thing to do is go to a trunk and press save, then press the red
    "You have made changes - when finished, click here to APPLY them" at the top

    Sparks...
     
    Sparks, Oct 26, 2005
    #9
  10. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    "Sparks" <postmaster@127.0.0.1> wrote in message
    news:435f6597$0$38039$...
    >> Hi Sparks,
    >>
    >> Here's my sip.conf - I can't see anything different (can you?)
    >>
    >> I've got all the same ports as you forwarded in the
    >> router.............but it still doesn't work :-(
    >>
    >> [general]
    >>
    >> port = 5060 ; Port to bind to (SIP is 5060)
    >> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    >> disallow=all
    >> externip=my.dyndns.address
    >> localnet=192.168.1.0/255.255.255.0
    >> allow=ulaw
    >> allow=alaw
    >> context = from-sip-external ; Send unknown SIP callers to this context
    >> callerid = Unknown
    >> canreinvite=no
    >>
    >> #include sip_nat.conf
    >> #include sip_custom.conf
    >> #include sip_additional.conf

    >
    > Looks fine to me!
    >
    > Have you reloaded the config files in asterisk since you made changed?
    > (Easiest thing to do is go to a trunk and press save, then press the red
    > "You have made changes - when finished, click here to APPLY them" at the
    > top
    >
    > Sparks...
    >


    Hmm, I have, yes.

    Perhaps you would be kind enough to post the relevent section of your
    sip_additional.conf file?

    Mine's:

    [locsipdisc]
    username=MyUsername
    type=peer
    secret=MyPassword
    qualify=yes
    nat=yes
    insecure=very
    host=sip.sipdiscount.com
    fromuser=MyUsername
    fromdomain=sipdiscount.com
    disallow=all
    authuser=MyUsername

    Cheers.
     
    Jono, Oct 26, 2005
    #10
  11. Sparks

    Sparks Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    >
    > Perhaps you would be kind enough to post the relevent section of your
    > sip_additional.conf file?
    >
    > Mine's:
    >
    > [locsipdisc]
    > username=MyUsername
    > type=peer
    > secret=MyPassword
    > qualify=yes
    > nat=yes
    > insecure=very
    > host=sip.sipdiscount.com
    > fromuser=MyUsername
    > fromdomain=sipdiscount.com
    > disallow=all
    > authuser=MyUsername
    >
    > Cheers.
    >


    I see your problem :)

    You have disallowed all codecs (disallow=all) but not allowed any!

    Mine is...

    [sipdiscount]
    username=MyUsername
    type=peer
    secret=MyPassword
    qualify=yes
    nat=yes
    insecure=very
    host=sip.sipdiscount.com
    fromuser=MyUsername
    fromdomain=sipdiscount.com
    disallow=all
    authuser=MyUsername
    allow=ulaw
    allow=alaw


    Sparks...
     
    Sparks, Oct 26, 2005
    #11
  12. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    Sparks wrote:
    ||| Perhaps you would be kind enough to post the relevent section of
    ||| your sip_additional.conf file?
    |||
    ||| Mine's:
    |||
    ||| [locsipdisc]
    ||| username=MyUsername
    ||| type=peer
    ||| secret=MyPassword
    ||| qualify=yes
    ||| nat=yes
    ||| insecure=very
    ||| host=sip.sipdiscount.com
    ||| fromuser=MyUsername
    ||| fromdomain=sipdiscount.com
    ||| disallow=all
    ||| authuser=MyUsername
    |||
    ||| Cheers.
    |||
    ||
    || I see your problem :)
    ||
    || You have disallowed all codecs (disallow=all) but not allowed any!
    ||
    || Mine is...
    ||
    || [sipdiscount]
    || username=MyUsername
    || type=peer
    || secret=MyPassword
    || qualify=yes
    || nat=yes
    || insecure=very
    || host=sip.sipdiscount.com
    || fromuser=MyUsername
    || fromdomain=sipdiscount.com
    || disallow=all
    || authuser=MyUsername
    || allow=ulaw
    || allow=alaw
    ||
    ||
    || Sparks...

    I'll give it a try.

    Thanks.
     
    Jono, Oct 26, 2005
    #12
  13. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    Sparks wrote:
    ||| Perhaps you would be kind enough to post the relevent section of
    ||| your sip_additional.conf file?
    |||
    ||| Mine's:
    |||
    ||| [locsipdisc]
    ||| username=MyUsername
    ||| type=peer
    ||| secret=MyPassword
    ||| qualify=yes
    ||| nat=yes
    ||| insecure=very
    ||| host=sip.sipdiscount.com
    ||| fromuser=MyUsername
    ||| fromdomain=sipdiscount.com
    ||| disallow=all
    ||| authuser=MyUsername
    |||
    ||| Cheers.
    |||
    ||
    || I see your problem :)
    ||
    || You have disallowed all codecs (disallow=all) but not allowed any!
    ||
    || Mine is...
    ||
    || [sipdiscount]
    || username=MyUsername
    || type=peer
    || secret=MyPassword
    || qualify=yes
    || nat=yes
    || insecure=very
    || host=sip.sipdiscount.com
    || fromuser=MyUsername
    || fromdomain=sipdiscount.com
    || disallow=all
    || authuser=MyUsername
    || allow=ulaw
    || allow=alaw
    ||
    ||
    || Sparks...

    That made a massive difference - I can now ring out, get the 0 Euro Cents
    message....But NO audio

    Grrrr

    BTW, why is disallow=all in there anyway?
     
    Jono, Oct 26, 2005
    #13
  14. Sparks

    Sparks Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)


    > That made a massive difference - I can now ring out, get the 0 Euro Cents
    > message....But NO audio


    .....bugger!

    not sure what to suggest, other than checking your routers settings!


    >
    > Grrrr
    >
    > BTW, why is disallow=all in there anyway?


    It is so you can then specify the codecs you want to use with that trunk -
    if for example your default was changed to GSM, then this trunk will stay
    with whatever you have specified.
     
    Sparks, Oct 26, 2005
    #14
  15. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    Sparks wrote:
    ||| That made a massive difference - I can now ring out, get the 0 Euro
    ||| Cents message....But NO audio
    ||
    || ....bugger!
    ||
    || not sure what to suggest, other than checking your routers settings!
    ||
    ||
    |||
    ||| Grrrr
    |||
    ||| BTW, why is disallow=all in there anyway?
    ||
    || It is so you can then specify the codecs you want to use with that
    || trunk - if for example your default was changed to GSM, then this
    || trunk will stay with whatever you have specified.

    Thanks for your help getting this far.

    I might be investing in a new router soon, though I can't see anything out
    of sorts there.
     
    Jono, Oct 26, 2005
    #15
  16. Sparks

    Jono Guest

    Re: VoIPDiscount with Asterisk - Success! (sipDiscount)

    Sparks wrote:
    ||| That made a massive difference - I can now ring out, get the 0 Euro
    ||| Cents message....But NO audio
    ||
    || ....bugger!
    ||
    || not sure what to suggest, other than checking your routers settings!
    ||
    ||
    |||
    ||| Grrrr
    |||
    ||| BTW, why is disallow=all in there anyway?
    ||
    || It is so you can then specify the codecs you want to use with that
    || trunk - if for example your default was changed to GSM, then this
    || trunk will stay with whatever you have specified.

    Well bugger me. I've finally got it working.

    I had to create an IAX2 trunk with the following settings:
    (note the use of Voipbuster's servers.)

    allow=ulaw&alaw
    authuser=MySipDiscountUsername
    disallow=all
    fromdomain=voipbuster.com
    fromuser=MySipDiscountUsername
    host=iax.voipbuster.com
    insecure=very
    nat=yes
    qualify=yes
    secret=MySipDiscountPassword
    type=peer
    username=MySipDiscountUsername
     
    Jono, Oct 26, 2005
    #16
  17. Sparks

    paul123 Guest

    Hi Sparks and all,

    I have a teething problem with Sipdiscount on Asterisk@home. I've used
    your settings in Asterisk@home (beta 2.1), but found I wasn't getting
    the account as "registered" when looking at the Asterisk info page...

    However, as I'd had a similar problem with voipbuster, I changed the
    lines:
    >>fromdomain=sipdiscount.com
    >>host=sip.sipdiscount.com

    to
    >>fromdomain=83.138.185.232
    >>host=213.61.187.146

    and it registers and works!....well, almost.

    I can make calls but I get a one way audio problem - I can't hear the
    person I'm calling, though they can hear me.

    Any ideas?

    Paul
     
    paul123, Oct 27, 2005
    #17
  18. Sparks

    sameerms18

    Joined:
    Nov 18, 2011
    Messages:
    1
    voipdiscount in elastix not working

    Hi,

    i have configured my voipdiscount account in elastix. i can call my local sip extensions. i am in saudi arabia now. i want to make outside calls through my voip provider trunk. what i need to do? i am new in elastix. i dont know how to configure dialing pattern for voipdiscount to make call to outside. please help me
     
    sameerms18, Nov 18, 2011
    #18
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