[Asterisk-Users] Inbound Free World Dialup - extension not ringing?

[Lyle Giese]

You need a defination for the inbound FWD and what to do with that.

In my extensions.conf, I have:

[globals]
FWDNUMBER=123456 ; your actual fwd number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010

[fwd_out]
exten => _123.,1,SetCallerId,${FWDCIDNAME}  ; replace 123 with the desired access code to dial out via FWD
exten => _123.,2,Dail(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN}:3},60,r)
exten => _123.,3,Congestion

[local]
include => fwd_out  :add to local context

[default]

;inbound dialing from FWD
exten => ${FWDNUMBER},1,Goto(housemenu,s,1)  ; I have mine set to hit a menu, no reason you cann't forward to an extension instead

  ----- Original Message ----- 
  From: Chris Blunt 
  To: asterisk-users at lists.digium.com 
  Sent: Sunday, August 15, 2004 3:29 AM
  Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?


   

  Hi to all the * people out there,

   

  Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast.

   

  My config is quite simple, I'm just following examples and the Wiki:  I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem).

   

  I have tried to set up Asterisk to accept calls from FWD on another number I have registered, but I can't get my local X-Lite to ring on an inbound call from FWD, and I get the busy tone on the BT100

   

  When I sip debug, I can see that I am registered with FWD, and when I call the number from the BT100 I can see all the incoming information but still nothing on my X-Lite.

   

  My extensions.conf:

   

   

  [general]

  static=yes

  writeprotect=no

   

  [globals]

   

   

  [sip]

  exten => 1,1,Dial(SIP/phone1,20,tr)

  exten => 2,1,Dial(SIP/phone2,20,tr)

  exten => 2,2,VoiceMail,u1234

  exten => 2,102,VoiceMail,b1234

  ;exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)

  exten => 1001,1,Ringing

  exten => 1001,2,Wait(2)

  exten => 1001,3,VoicemailMain,s1234

  exten => 6601,1,WaitMusicOnHold(60)

  exten => 232999,1,Dial(SIP/phone1,30,tr)

  exten => 232999,2,Hangup

   

   

  I am behind a NATed fire wall, but I'm not sure that is related.

   

  Any ideas or help (working simple confs) would be much appreciated.

   

   

   

  Best regards

   

  --

   

  Chris Blunt

   

  SIP: 248189 at fwd.pulver.com

   

   

   
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040815/1c7fc6e6/attachment.htm