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

[Lyle Giese]

I have my server on a public IP address, so I am not behind any NAT.

But take a look at http://www.voip-info.org/wiki-Asterisk+config+iax.conf

One of the problems could be the notransfer option in this config file.  I suspect that what you may be seeing(and I am no expert, but have been reading here extensively) is FWD and Asterisk trying to drop out of the voice path by having the two 'phones' talk directly to each other and are unable to accomplish this because of the NAT.  The canreinvite and nat options control this behavior for sip.

Lyle

  ----- Original Message ----- 
  From: Chris Blunt 
  To: asterisk-users at lists.digium.com 
  Sent: Sunday, August 15, 2004 5:06 PM
  Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?


  Hi Lyle, 

   

  Thank you so much for your help, I think your information points to using IAX2 rather than registering with FWD from the sip.conf

   

  I have made an attempt to understand this, added the appropriate information into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX registration box, and I now get my local sip phone ringing when I dial in from FWD!   Hurrah, unfortunately I get no sound in either direction.  Do you have any experience of this or could it be due to me being inside a NAT firewall?  I have port 5060 forwarded to my * server, should I forward any other ports? (I can only forward a maximum 20 single ports due to a limitation on my home router).

   

  As yet I am unable to make outgoing calls over FWD, I figured I would look at this next.

   

  Is there a NAT solution that could be used with sip.conf rather than the IAX?

   

  Again your help is most appreciated.

   

  Best regards

   

  Chris

   


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

  From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Lyle Giese
  Sent: 15 August 2004 15:14
  To: asterisk-users at lists.digium.com
  Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

   

  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

     

     

     
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