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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Lyle Giese
Sent: 15 August 2004 15:14
To: [EMAIL PROTECTED]
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/${FWDNUMBER}:[EMAIL PROTECTED]/${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

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: [EMAIL PROTECTED]

 

 

 

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