Hi Dave,
Still not working with IAX after I added nat=yes and externip=registered
IPfrom rogers to iax.conf
It looks like the very first request came thru ok but the *server was
not able to send response back to the phone. How does the *server know
which IP it should respond to??
Any suggestion? I am getting so frustrated...
Please note that the IPs and host names have been edited.
Thanks,
Richard
Below is the unsuccessful request from the softphone on the internet to
*server via firewall on port 4569
00:13:33.507916 IP
CPE01235e967998c09-CM90012g90fdd003dd60036a90.cpe.net.cable.rogers.com.6
0201 > 192.168.0.111.4569: UDP, length 21
Below is the successful handshake inside the firewall. 192.168.0.111 is
my *server
00:03:51.346253 IP 192.168.0.101.4569 > 192.168.0.111.4569: UDP, length
21
00:03:51.346515 IP 192.168.0.111.4569 > 192.168.0.101.4569: UDP, length
32
00:03:51.351558 IP 192.168.0.101.4569 > 192.168.0.111.4569: UDP, length
51
-- Registered IAX2 '838' (AUTHENTICATED) at 192.168.0.101:4569
00:03:51.356104 IP 192.168.0.111.4569 > 192.168.0.101.4569: UDP, length
12
00:03:51.356148 IP 192.168.0.111.4569 > 192.168.0.101.4569: UDP, length
58
00:03:51.366608 IP 192.168.0.101.4569 > 192.168.0.111.4569: UDP, length
12
00:03:51.366776 IP 192.168.0.101.4569 > 192.168.0.111.4569: UDP, length
18
00:03:51.366871 IP 192.168.0.111.4569 > 192.168.0.101.4569: UDP, length
12
00:03:51.371579 IP 192.168.0.101.4569 > 192.168.0.111.4569: UDP, length
12
Thanks,
Richard
-----Original Message-----
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 31, 2007 2:50 PM
To: [EMAIL PROTECTED]
Cc: [email protected]
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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> > > >
> > > >
> > >
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