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 <[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 > > > > > > > > > > > > > > > > > > > > > -- > > Bill Chmura > Director of Internet Technology > Explosivo ITG > Wolcott, CT > > p: 860.621.8693 > e: [EMAIL PROTECTED] > w. http://www.explosivo.com
-- 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
