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Just in case other people on the list have this problem... Begin forwarded message: > From: "George Lin" <glin@cosini.com> > Date: Thu Aug 14, 2003 6:54:46 AM Europe/Budapest > To: "Paul Cheng" <asterisk@klarium.com> > Subject: RE: FW: [Asterisk-Users] SIP NAT question > > Dear Paul, > > Thanks for the suggestion. It works now. > > Thank you very much. > > George Lin > > -----Original Message----- > From: Paul Cheng [mailto:asterisk@klarium.com] > Sent: Wednesday,...

I asked on the IRC channel last night and was told the IAXTEL had been down for a few months now. It had a very poor uptime.. Maybe someone can tell us why the uptime was so poor. Alex Message: 9 Date: Wed, 9 Jul 2003 01:05:00 +0200 From: Paul Cheng <asterisk@klarium.com> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IAXTEL toll-free Reply-To: asterisk-users@lists.digium.com Hi, Has anyone been able to place a call via IAXTEL toll-free termination lately? I had it working at one time, but now it doesn't seem to work anymore. ww...

...>PC which kills a major cost saving for going VoIP.. > >Later.. > > >--__--__-- > >Message: 10 >Date: Wed, 15 Oct 2003 08:56:42 +0200 >Subject: Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec >From: Paul Cheng <asterisk@klarium.com> >To: asterisk-users@lists.digium.com >Reply-To: asterisk-users@lists.digium.com > >Our experience with the Budget Tones 101have been poor as well. The >devices seem to die after a day or two (even with the new firmware) and >then need to be rebooted. On occasion, the dev...

Hi, I recall seeing a sample extensions.conf file that allowed tollfree calls to be routed via iaxtel to the US and the NL, but I must be going blind, because I've scoured the list but can't find it. Can someone send it to me if they have it? Much appreciated. Thanks! --- Paul Cheng M?ty?s kir?ly ut 10 H-1121 Budapest HUNGARY paul.cheng@alum.mit.edu mobile: +36 30 381-9311

Hi, Does anyone know the difference between RFC2543 and RFC3261? They are both SIP, but apparently incompatible. We are testing some hardware devices that support RFC3261 and it appears Asterisk is supporting RFC2543 and not completely compatible with RFC3261 (see below). Does anyone know how to configure Asterisk so it can do what is missing? 3. The problem occurs with confirm message

Hi, I searched the archives about this, but didn't find any references. When I make an outbound SIP call, the call completes and everything is fine, but in the Asterisk console, I keep getting a huge stream of warning messages: "WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames" I thought I saw this in a post earlier, but I

Hi, I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use g.723) Asterisk will connect to iConnect, successfully natively bridge the call and then about two seconds later not just drop the call, but terminate unexpectedly.

Does anyone know what this means? I suspect it has something to do with ztdummy as app_meetme no longer works--though it did a month or two ago.

Hi, I've seen some postings on the Directory application, but haven't seen too many resolution postings. Has anyone experienced where the Directory app doesn't even answer when called? For example, using the config below, dialing 899 results in just a continual ringing sound. extensions.conf exten => 899,1,Directory(local) exten => 899,2,Hangup [local] exten =>