Richard:

You've spent more than 48 hrs trying to find a solution for this issue.  In my 
personal and professional experience, finding the exact root of a problem has 
usually been over 90% of the time invested behind an issue, and less than 10%  
time invested in actually implementing the solution that makes things work.    

We are all relying on your input via e-mail to help you trouble shoot.    If 
you allow me, or Simon or other Asterisk experts to log into your Asterisk box, 
chances are very high we will find the exact cause of your problems.  When we 
have identified the cause, we can quickly provide the solution.

Of course we are not cheap and have our hourly rates - but I am willing to make 
an exception to help resolve the issue without cost, as long as you treat the 
rest of the folks here who has tried to help you, with some good beer and me a 
glass of root beer since I have a high intolerance for alcohol.   

I'm normally available after 9:00 PM to engage in non-paying work.  

Cheers!
Reza.


----- Original Message ----- 
From: "Richard (Rogers @ work)" <[EMAIL PROTECTED]>
To: "'David Cook'" <[EMAIL PROTECTED]>
Cc: <[email protected]>
Sent: Saturday, June 02, 2007 12:27 AM
Subject: RE: Re: [on-asterisk] Allow inbound SIP traffic thru d-link router


> 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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>> > >
>> > >
>> > >
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>> >
>> >
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>>
>>
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> 
> ----- End forwarded message -----
> 
> 
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