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...pe this may help you, I would take the project, but I have no time righ now. Maybe in about 1.5 months, when my degree work is done. c-ya! Ildefonso Camargo icamargo@unet.edu.ve ildefonso_camargo@yahoo.com >Message: 3 >Date: Tue, 19 Aug 2003 13:35:53 -0500 (CDT) >From: James Sharp <jsharp@psychoses.org> >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Oen source IP phone, maybe? >Reply-To: asterisk-users@lists.digium.com > > >Its another one of my "If I only had time...damn this sleep thing" ideas, >but I really wonder how hard/cost...

...I have no time > righ now. Maybe in about 1.5 months, when my degree work is done. > > c-ya! > > Ildefonso Camargo > icamargo@unet.edu.ve > ildefonso_camargo@yahoo.com > >> Message: 3 >> Date: Tue, 19 Aug 2003 13:35:53 -0500 (CDT) >> From: James Sharp <jsharp@psychoses.org> >> To: asterisk-users@lists.digium.com >> Subject: [Asterisk-Users] Oen source IP phone, maybe? >> Reply-To: asterisk-users@lists.digium.com >> >> >> Its another one of my "If I only had time...damn this sleep thing" >> ideas, b...

Will M$ ever stop!!.. Whats the bet their telecoms products will use non-standard protocols.. I really wouldn't like to run a telecom system on Windoze in the first place.. Full Story.. http://news.zdnet.co.uk/communications/0,39020336,39117099,00.htm

Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo

We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2

Are any signals (SIGTERM, SIGINT, SIGDIERIGHTNOW?) passed to an AGI script if it gets hung up upon before it exits?

Its another one of my "If I only had time...damn this sleep thing" ideas, but I really wonder how hard/cost effective it would be to build an open source IP phone or phone adapter (ala ATA). In about 20 minutes of mulling and research, I figure you could do it for about $40 in parts plus coding time... 1 DS80C400 Ethernet enabled microcontroller with built in IPV4/V6 stack $10 1

If I have a system with 1 machine to handle incoming H.323 calls and then multiple machines to distribute them to T1 ports over TDMoE, where does the codec translation take place? Does it take place in the master system or does it take place in each of the slave TDMoE systems? Also, any idea how many concurrent G.729 calls a system like this would handle?

Is the max recommended still 2 cards, even in a Quad Xeon with superduperwhizbang Hyperthreading? I'll be running incoming G.729 audio out to TDM.

I've got a handful of T1s going into a TE410. When I place calls into the system over these T1s, the system either doesn't decode all of the DTMF digits or it decodes ones that aren't there. When the system places calls out, there is no problem doing the DTMF detection. Everything works great.

Do they fall under FCC certification if they're built to the same specifications as the ones from Digium? If I build my own T100Ps from the schematics and board layouts that are available, are they legal to plug into the PSTN?

I've got a single phone line coming into an X100P. In extensions.conf I've got this: [inboundzap] exten => s,1,Answer exten => s,2,EAgi,hanguptest.agi I see the ring come in and Asterisk detects it and tries to do something with it: NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer("Zap/1-1", "") in

I was looking thru the archives and did not see anything, might not have looked far enough back. I installed Asterisk@Home 0.7 today, and connected it to an analog port of our phone system. Using extension 200 and SIPphone lite I can get incoming calls, problem is reaching the other extensions on the phone system that I am connected to and also dialing outgoing calls. The phone system

Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris & Linux explode when loading login_page.php.

I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the