[Asterisk-Users] PLease help: Asterisk to Quintum interconnection

[Jessie V. Mabanglo]

Hi Pulu 'Anau,

thank you very much for the advice.. we already did some part.. and still we are on simulation to prefect it...

thank you very much again..

Regards,

Jessie
  ----- Original Message ----- 
  From: Pulu 'Anau 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, February 23, 2005 5:55 PM
  Subject: Re: [Asterisk-Users] PLease help: Asterisk to Quintum interconnection


  I have an old A800 with the newer sip firmware that I've been using for quite a while.  They lose a bit being put into SIP (most noticeably the ability to send dtmf out of band) but I had problems with chan_h323 a while ago and got tired of all the hoops for oh_323 so was happy to switch.

  I use the builtin dialplan as little as possible. As soon as someone picks up the phone it goes straight to asterisk.  This also means that I can just use one hunt number for all outgoing pstn calls, even port to port it goes through the * server.


  For the sip settings, I just use the proxy, not the registrar, as I said all calls go straight to * anyway.  

  Here's one pstn trunk group:
  Name = 27946-7
  Pass Through = no(0)
  Provide Call Progress Tone = no(0)
  Busyout = no(0)
  Hunt Algorithm = ascending(0)
  Modem Bypass = no(0)
  Direction = both(2)
  DN Used = public
  Forced IP Routing # = 1000
  Forced IP Routing # Type = public
  IP Extension = yes(1)
          channel      ip-addr         dnis     rmt-line chan
  Maximum LAM Calls Allowed = 8
  LAM: Index Pattern Replacement NumberType
           1 <           9> <            > 0

  That's only the stuff I changed or think is really that important.  The forced ip routing means it answers the phone immediately on pstn and dials ext 1000 on the asterisk machine.  The big thing is the lam pattern stuff.  You have to put in a pattern for the quintum to match, otherwise it will give sip errors, as it doesn't understand where to send any incoming calls.  It could be anything, but I just started with a 9, the only thing is it can't start with the same thing as any of the extensions that're on the pbx side.

  On the pbx side:
  Name = 711
  Pass Through = no(0)
  Hunt Algorithm = ascending(0)
  Direction = both(2)
  DN Used = public
  Forced IP Routing # = 1000
  Forced IP Routing # Type = public
  IP Extension = yes(1)
          channel      ip-addr         dnis     rmt-line chan
  Public Number of Digits = 3
  Public Hunt Ldn's:
           1: 711

  Pretty much the same.  You'll see the pub hunt ldn which is the extension that I dial from asterisk (see the extensions.conf below).  This also goes straight to * which means there's a bit of a delay when you pick up the phone - not enough to notice but if you pick up the phone and dial straight away it might not catch the first digit.  The caller id gets set to "Quintum" <name> which is the name of the pbxtg, which is why it's set to the extension.  

  Anyway on the asterisk side the sip.conf is pretty basic, but make sure you have dtmf=inband.

  Some parts of my extensions.conf:

  exten => _71X,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}@tenor800,30)

  Dials the pbx side...  tenor800 is the name from the sip.conf

  exten => _9XXXXX,1,Dial(SIP/${EXTEN}@tenor800,30|mH)

  Dials out on the pstn... As you can tell I start extensions going out on the local pstn with a 9 in asterisk as well... If you just dial them straight you'll have to add the 9 before the exten variable.

  Hope that helps... I don't imagine that it does any kind of authentication on calls coming into it but since mines natted behind two firewalls on a lan with the * machine I've never really checked.  

  Pulu



--
Pulu 'Anau
27946 x 711
878-7856




  Jessie V. Mabanglo wrote: 
    My fellows,

    We have Asterisk at home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that?

    Your time is very much appreciated..

    Cheers,

    Jessie
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