[asterisk-users] Asterisk 1.4 and Cepstral

2007-09-17 Thread Russell Handorf
Greetings, I've recently upgraded from Asterisk 1.2 to 1.4. I've been searching for a solution, but am also trying the easy way at the same time. I've now got David of Cepstral now speaking using app_swift from http://www.mezzo.net/asterisk/app_swift.html . The problem is, he sounds way worse

Re: [asterisk-users] 99 bottles of beer

2007-08-22 Thread Russell Handorf
I've been working on an X10 component already. It works, but I wish the CMA15 would work correctly in Linux (I know it's suppose to, but for whatever reason the one I have just doesnt.) It's just a little AGI script that I have working with Cepstral that throws http PUTs to the Windows box that

Re: [asterisk-users] Dial plan suggestions

2007-08-14 Thread Russell Handorf
Here's some details for you all. Asterisk 1.2 Polycom 301/601 phones As for my existing dial plan, I'm considering starting from scratch. Thanks again. Gerald A wrote: > Hiya, > > On 8/14/07, *Russell Handorf* <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>

[asterisk-users] Dial plan suggestions

2007-08-14 Thread Russell Handorf
Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). He

[Asterisk-Users] app_flite.so

2006-06-06 Thread Russell Handorf
Hello all, I'm playing with app_flite, as I'm sure you've guessed. I have the sources compiled and running, headers and libraries in their respective places. I then compiled app_flite without any problems or errors. When I try to have asterisk execute the module, I get the following error Ju

Re: [Asterisk-Users] Anyone know who is in this picture?

2005-11-19 Thread Russell Handorf
http://www.walnutfactory.org/truth.shtml Greg Boehnlein wrote: On Fri, 18 Nov 2005 [EMAIL PROTECTED] wrote: I couldn't find his bio on rotten.com http://www.rotten.com/library/bio/hackers/captain-crunch/ ___ --Bandwidth and Coloca

Re: [Asterisk-Users] TDM01B

2005-03-27 Thread Russell Handorf
Does someone have a working config file they could send me? Thanks r Steven Critchfield wrote: On Sun, 2005-03-27 at 16:00 -0600, Rich Adamson wrote: Might try modprobe zaptel then modprobe wcfxo (or wctdm). The order makes a difference and I don't remember exactly which one comes first. Mo

Re: [Asterisk-Users] TDM01B

2005-03-27 Thread Russell Handorf
guess my hunch of the system not seeing the card might be the right track. administrator tootai wrote: Russell Handorf a écrit : Greetings all, I addressed the issue below with editing the .config file in my 2.6.10 source file by adding CONFIG_CRC_CCITT=m recompiled, and now I can modprobe wcfx

Re: [Asterisk-Users] TDM01B

2005-03-26 Thread Russell Handorf
Greetings all, I addressed the issue below with editing the .config file in my 2.6.10 source file by adding CONFIG_CRC_CCITT=m recompiled, and now I can modprobe wcfxs and zaptel without any errors, and ztctl -vvv shows Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. All

Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage

2005-03-22 Thread Russell Handorf
I was only able to get the softphone account to make inbound calls on one sip.conf config, or outbound calls on another sip.conf config. I didnt investigate the issue completely, but from what I could tell, they wouldnt allow multiple SIP sessions from the same IP address. I didnt try running 2

Re: [Asterisk-Users] Net2Phone / Vonage

2005-03-22 Thread Russell Handorf
Just got off the phone with Net2Phone; they now require 3 credentials to authenticate: account id, pin number, and MAC address. Any ideas? Thanks Russell Handorf wrote: I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after

Re: [Asterisk-Users] Net2Phone / Vonage

2005-03-21 Thread Russell Handorf
I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication

Re: [Asterisk-Users] US pstn => voip

2005-03-21 Thread Russell Handorf
voicepulse? We get free inbound on them. However, every once in a while the service degrades for quite some time and they blame it on their upstream provider; the issue just "goes away" without any real resolution. Mark Charlton wrote: Hi I believe this is due to the way US phone systems work, h

Re: [Asterisk-Users] Net2Phone / Vonage

2005-03-21 Thread Russell Handorf
secure = very canreinvite = yes context = Home Anyone who is using n2p at all? Thanks Russell Handorf wrote: Greetings all, I've got Net2Phone and Vonage pitting against each other right now. At the moment, with the Vonage's Softphone account, I can only make incoming or outgoing phone calls (

[Asterisk-Users] Net2Phone / Vonage

2005-03-20 Thread Russell Handorf
Greetings all, I've got Net2Phone and Vonage pitting against each other right now. At the moment, with the Vonage's Softphone account, I can only make incoming or outgoing phone calls (one config for incoming, one for out); in otherwords I cant seem to have one sip.conf file that will allow ast