[Asterisk-Users] Re: Help with Gizmo from outside firewall

[Bill]

Sorry, send this part from an unregistered account

> 
> I know this is going to a "duh" statement to a lot of people, but just
> in case... when the non-audio gizmo connection rolls to voicemail, on
> the cli I get:
> 
>  app.c:645 ast_play_and_record: No audio available on
> SIP/proxy01.sipphone.com-xxxxxxxxx??
> 
> I am guessing this is since there is no RTP connection.
> 
> Thanks
> 
> Bill
> 
> 
> 
> 
> 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
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> 
> 
> -- 
> 
> Bill Chmura
> Director of Internet Technology
> Explosivo ITG
> Wolcott, CT
> 
> p: 860.621.8693
> e: bill at Explosivo.com
> w. http://www.explosivo.com


-- 

Bill Chmura
Director of Internet Technology
Explosivo ITG
Wolcott, CT

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