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@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
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@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@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@Explosivo.com > w. http://www.explosivo.com-- Bill Chmura Director of Internet Technology Explosivo ITG Wolcott, CT p: 860.621.8693 e: bill@Explosivo.com w. http://www.explosivo.com
Bill
2006-Mar-15 18:29 UTC
[Asterisk-Users] Help with Gizmo from outside firewall <- update
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@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@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 > > > > > > > > > > _______________________________________________ > --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: bill@Explosivo.com w. http://www.explosivo.com