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

[Edward Eastman]

IAX2 uses udp port 4569, so you’ll probably have to open that up on your
firewall/router.

http://www.voip-info.org/ is a good starting place for any asterisk problems
- specifically:

http://www.voip-info.org/wiki-Asterisk+firewall+rules
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD

HTH

Ed

________________________________________
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Chris Blunt
Sent: 15 August 2004 23:06
To: asterisk-users at lists.digium.com
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