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 <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > > [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 > > > > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Bill Chmura Director of Internet Technology Explosivo ITG Wolcott, CT p: 860.621.8693 e: [EMAIL PROTECTED] w. http://www.explosivo.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
