[Asterisk-Users] Help with Gizmo from outside firewall <- update

[Bill]

Well, I got off site today with my notebook and an x-lite install.  I
was able to connect into to the system and hear things, etc...

But since the phone connects ahead, this may be a different thing than
an incoming gizmo call eh?

If someone could even point me in the direction to look, I would be
greatful!




On Wed, 15 Mar 2006 15:06:47 -0500
Bill <Bill at explosivo.com> spake:

> 
> I've beaten myself bloody dealing with this one...  No luck so far.  In
> summary, incoming calls from Gizmo establish, but neither get nor send
> sound.  Outbound calls to Gizmo work fine (well a bit choppy but work)
> 
> My thought is that the SIP connection is being made fine, but the RTP
> is getting stopped / blocked / misdone somewhere.
> 
> Here is the thing:
> 
>  Asterisk 2.5 on Linux
>  (No hardware cards yet)
>  X-Lite softphones on a few machines
>  Gizmo clients and Gizmo accounts on the internet
>  Gizmo client on the localnet
>  PF firewall
>  New to asterisk
> 
> Okay - here are things that work and what I have tried:
> 
> Works:  If I call a Gizmo user outside the network from an XLite SIP
> phone inside the network it works.
> 
> Works:  If I call a Gizmo user inside the network from an XLite phone
> inside the network it works.
> 
> NOT WORK:  If I have asterisk register with gizmo and a gizmo person
> outside the network calls me, they get connected - but no sound either
> way.
> 
> NOT WORK:  If I have gizmo inside my network and I dial to my asterisk
> connected gizmo line it connects, but no sound.
> 
> I logged all dropped packets at the firewall and am not blocking
> anything (I was at first dropping some incoming UDP in the 9000-20000
> range, but that has been fixed.
> 
> The only thing I have not been able to do is to try to have an external
> xlite phone connect in and work.  I think this would rest the blame on
> the firewall or gizmo... 
> 
> The only thing that seems weird is that is only happens when Gizmo
> originates the call.  I can see the prompts and stuff playing on the
> CLI, but nothing gets sent to the other end.  Also, if I answer a call,
> sound goes neither way.
> 
> 
> I've tried a bunch of things
> My SIP.conf has
> 
> register => 1747xxxxxxx:password at proxy01.sipphone.com
> 
> [gizmo-inbound]
> type=peer
> context=from-gizmo
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> allow=alaw
> allow=ilbc
> allow=gsm
> nat=yes
> host=proxy01.sipphone.com
> insecure=very
> canreinvite=no
> externip=69.10.14.12
> localnet=192.168.0.0/255.255.255.0
> 
> I have no idea what to check / try next...  My gut instinct tells me it
> has to do with the firewall NAT and the RTP connection - but nothing is
> getting dropped or blocked, and I can dial out to them.  
> 
> Internally, Xlite -> asterisk works fine also.
> 
> Any ideas would be immense help!
> 
> 
> Bill
> 
> 
> 
> 
> 
> 
> 
> 
> 
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-- 

Bill Chmura
Director of Internet Technology
Explosivo ITG
Wolcott, CT

p: 860.621.8693
e: bill at Explosivo.com
w. http://www.explosivo.com