Hi Dave,

I am not sure how you mean by _not_ restrict ip but I have host=dynamic as
follows.
Can you see any problem with my config?
Thanks,
Richard

[general]
bindport = 4569           ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes


[999]
type=friend
setvar=REALCALLERIDNUM=999
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=4569
notransfer=yes
host=dynamic
dial=IAX2/999
context=from-internal
callerid=device <999>

----- Original Message ----- 
From: "David Cook" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[email protected]>
Sent: Thursday, May 31, 2007 2:50 PM
Subject: Fwd: Re: [on-asterisk] Allow inbound SIP traffic thru d-link router


> I might have been a bit quick of the blocks here.
>
> In /etc/asterisk/iax.conf your definition for the softphone will need to
>  _not_ restrict ip address ranges that your softphone is connecting
> from, it also should have host=dynamic.
>
> Bring up an ssh session with your server and load the asterisk cli while
> you launch the softphone. If you see nothing on the cli, then your
> forwarding is probably to blame. If you see something, it probably will
> tell you why you are failing to register and you can take action as
> appropriate.
>
> dbc.
> --
> David Cook
> ----- Forwarded message from David Cook <[EMAIL PROTECTED]> -----
>     Date: Thu, 31 May 2007 14:39:56 -0400
>     From: David Cook <[EMAIL PROTECTED]>
> Reply-To: David Cook <[EMAIL PROTECTED]>
>  Subject: Re: [on-asterisk] Allow inbound SIP traffic thru d-link router
>       To: "Richard (Rogers @ work)" <[EMAIL PROTECTED]>
>
> That infers you don't have 4569/UDP on your d-link box forwarded to your
> asterisk server.
> --
> David Cook
>
>
> Quoting "Richard (Rogers @ work)" <[EMAIL PROTECTED]>:
>
> > Thanks Dave.  My problem is pretty wier.  I know my IAX trunk works
> > but my
> > IAX softphone does not work from the internet.  It does not even
> > register
> > with my *server.  It only works internally.
> >
> > Richard
> > ----- Original Message -----
> > From: "David Cook" <[EMAIL PROTECTED]>
> > To: "Richard (Rogers @ work)" <[EMAIL PROTECTED]>
> > Cc: <[email protected]>
> > Sent: Thursday, May 31, 2007 1:38 PM
> > Subject: Re: [on-asterisk] Allow inbound SIP traffic thru d-link
> > router
> >
> >
> > > IAX will require port fwd 4569/UDP from the outside to your
> > Asterisk
> > > box. Control & data use the same port so no nat issues.
> > > --
> > > David Cook
> > >
> > >
> > > Quoting "Richard (Rogers @ work)" <[EMAIL PROTECTED]>:
> > >
> > > > Thanks Simon.  Yes, my network was what you suggested.
> > > > I will make the additional config on my *server tonight and try
> > it.
> > > >
> > > > On the same token, can you suggest any additional config required
> > for
> > > > IAX2?
> > > > I am also having problem with IAX2 traffic too.
> > > >
> > > > Thanks a lot!
> > > > Richard
> > > > ----- Original Message -----
> > > > From: "Simon P. Ditner" <[EMAIL PROTECTED]>
> > > > To: "Richard (Rogers @ work)" <[EMAIL PROTECTED]>
> > > > Cc: <[email protected]>
> > > > Sent: Thursday, May 31, 2007 11:54 AM
> > > > Subject: Re: [on-asterisk] Allow inbound SIP traffic thru d-link
> > > > router
> > > >
> > > >
> > > > > Is this what your network looks like?
> > > > >
> > > > >  [Asterisk] -- [Dlink Router] -- (Internet) -- [Phone] ?
> > > > >
> > > > > On the Asterisk server, you will need the following in your
> > > > sip.conf:
> > > > >
> > > > >  externip = <public IP address of your connection>
> > > > >
> > > > > or:
> > > > >
> > > > >  externhost=<DNS name, i.e. home.dyndns.org>
> > > > >  externrefresh=10
> > > > >
> > > > > You might also need to put the following in your phone's
> > profile:
> > > > >
> > > > >  [phone1]
> > > > >  nat=yes
> > > > >
> > > > > In rtp.conf, you should see a range of UDP ports that should
> > likely
> > > > be
> > > > > forwarded to your asterisk box in your D-Link router as well.
> > i.e.:
> > > > >
> > > > >  rtpstart=10000
> > > > >  rtpend=20000
> > > > >
> > > > > Cheers,
> > > > > spd
> > > > >
> > > > > On Thu, 31 May 2007, Richard (Rogers @ work) wrote:
> > > > >
> > > > > > Hi,
> > > > > >
> > > > > > I know there was a session on SIP last night but I was unable
> > to
> > > > attend.
> > > > I
> > > > > > am hoping some of you could help me out based on what you
> > > > learned.
> > > > > > I am having problem getting my router to work with my Sip
> > phone
> > > > which is
> > > > > > connecting from the internet to my asterisk server.
> > > > > >
> > > > > > I am not able to get the voice channel thru my router.  And
> > my
> > > > SIP phone
> > > > > > sometimes register with the server successfully but sometime
> > not.
> > > > > > I opened up both TCP and UDP port 5060 to point to my *
> > server.
> > > > > >
> > > > > > Is there anything else I have to add?  I am not sure how I
> > would
> > > > add a
> > > > range
> > > > > > of ports to my * server.
> > > > > >
> > > > > > Thanks,
> > > > > > Richard
> > > > > >
> > > > > >
> > > > > >
> > > >
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