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...oicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: Todd Lieberman [mailto:todd@tlsolutions.net] Sent: Monday, November 24, 2003 9:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Pressing 0 in Voicemail causes * to hangup The Problem: When a call gets into voicemail from Queue and presses 0 before leaving a message * will issue a Hangup. I...

...on managing the list at > asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. RE: voicemail file access problems (Todd Lieberman) > 2. sip -> h323 -> ptsn (Brian West) > 3. RE: voicemail file access problems (Todd Lieberman) > 4. Re: voicemail file access problems (Tilghman Lesher) > 5. Re: sip -> h323 -> ptsn (Patrick) > 6. RE: voicemail file access problems (Patrick) > 7. Re:...

...ty, but it works great. The nice part about IAX is that it is so obscure that most hackers will ignore it as they want to crack SIP or H323. IAX is pretty safe. Regards, TL -----Original Message----- From: lists [mailto:lists@efastfunding.com] Sent: Tuesday, May 18, 2004 10:43 PM To: 'Todd Lieberman' Subject: RE: [Asterisk-Users] * and Cisco routers The problem is I have 50-60 7940 just sitting in very large closet and I also have about 150 DIDs free. The other thing is I would have a very very hard time justifying new hardware. I am currently deploying cisco's Ip softphones which...

...hat is not working... I tried like you mentioned it and even a few different ways and will not create the file at all.... Am I doing something wrong... -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Todd Lieberman Posted At: Sunday, May 09, 2004 4:00 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] AGI Assistance Subject: RE: [Asterisk-Users] AGI Assistance Declare the file path before you record it. $path = "/usr/local/apache/htdocs/demo/sound/myapp.$date.wav"; $AGI->exec(...

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Hello, I have researched a few postings where users mentioned being able to install Asterisk on Mac OS X Panther by adding some code after line 165 in the Makefile and then compiling. This has been unsuccessful for me. I downloaded the asterisk-0.9.0.tar.gz tarball and am trying to install from it. The output I get upon trying to make after editing the Makefile can be viewed at:

...sh) > 2. Re: Booting error - Unable to specify channel > 2: No such device (Anon) > 3. newbie - Asterix and modem cards (Alex G > Robertson) > 4. Re: newbie - Asterix and modem cards (Anon) > 5. RE: Booting error - Unable to specify channel > 2: No such device (Todd Lieberman) > 6. RE: vm e-mail notification stopped (Uriel > Carrasquilla) > 7. Re: Booting error - Unable to specify channel > 2: No such device (Anon) > 8. Re: Adding two FXO cards - not working > (Hermann Wecke) > 9. editing errors/typos in rev 2 of The Asterisk > Handbo...

I have a couple of questions I'm hoping folks can help me out with. When I search the mailing list, I see folks doing what I'm interested in so here's hoping ! - How are people making out with interfacing to the 7960? I'm considering buying a number of these as they look quite feature rich. But, are they easy to interface to? - Will I be able to interface with softkeys on

I'm having problems with a new install of Asterisk (I had to reinstall because hard drive failed). I've used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for

...;conf" key and then dial the next extension. The new call connects and I hit "conf" again but the calls do not get bridged. Any Suggestions? I'm using the config files from http://www.loligo.com/asterisk/Cisco/79xx/2003-06-20.from-ftpeng.cisco.c om/ Thanks, TL -- Todd Lieberman 800-675-3192 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030714/0409754f/attachment.htm

Hi, When I use the analog phone connected to Zap/1 how do I transfer & hold the caller ? When I hit the flash key, all that happens is the caller hears a beep (sounds like DTMF). But no stutter dial tone on the Zap/1 Port, just continuing conversation with the caller. What could be wrong here? Cheers Jay

How do I stop asterisk when it is in a bad mood? It keeps dialing extensions and won't listen! I tried kill <PID>. No go. I don't want to have to reboot again. Thanks. Jim Friedeck P.S. I love it when my boss looks over my shoulder and I don't have an answer when he says: 'So, what are you doing?'

A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389

Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L

Hello Everybody, Is there a function available for calculating the density of a non-central t distribution? (dt does not accept the ncp option). Thanks in advance! -- Wolfgang Viechtbauer "Are you not thinking what I am not thinking?" -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read

Hi, Is it possible to stop users from having the option to change their voicemail greetings ? I'm running asterisk on an old machine with a small hard drive and I don't want users to be able to take up room on it by recording their own messages. Also, once someone has recorded their own messages, cant it be set back to the default one ? Once final question, is it possible to see how

Anyone successfully implemented a solution for allowing ZapBarge call monitoring only for a specific group of agents calls? The issue I see is that the feature only works on zap channels, and all of the agents (in many cases) are IP phones. Allowing ZapBarge and ZapScan on the TDM PSTN (t100p) interface has privacy issues for senior managers, but would allow all outbound zap calls to be

Is there is an open source ACD component for Asterisk? Likewise, is there an open source CTI component that will work with Asterisk? Regards, Jim O'Brien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/e6ee98da/attachment.htm

FYI, The LLVM Performance Workshop will be held at CGO 2017. The workshop is co-located with CC, HPCA, and PPoPP. If you are interested in attending the workshop, please register at the CGO website: http://cgo.org/cgo2017/workshops.html When: Saturday February 4th, 2017 Where: Austin, Texas, USA ---- Hi, CGO workshop and tutorial organizers, This is a friendly reminder that early

Here are a few things I would like to see .. 1. In addition to time/date stamps, store/read the caller id info with the voicemail messages. 2. Have the ability to configure the system to ignore and delete messages left by a caller that are 3 seconds or less (maybe make this configurable) Not sure but that would cut alot of hangup calls out of your voicemail box. I can't think of much more