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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

...t but what the hell do you do with 50-60 7940's that we paid 315 or less per? Doug Block Chief Information Officer of Efast Funding 713-983-4055 (Direct) 888-338-3863 x 4055 (Toll Free) 713-983-4555 (Direct Fax) 832-483-4495 (Cell) -----Original Message----- From: Todd Lieberman [mailto:todd@tlsolutions.net] Sent: Tuesday, May 18, 2004 8:23 PM To: lists@efastfunding.com Subject: RE: [Asterisk-Users] * and Cisco routers Hi Doug, What your talking about is setting up 50-60 home networks, it' will consume your time. I suggest the IAXy ATA with the IAX2 protocol or some other NAT friendly...

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, 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:

...9;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'm sure it's...

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

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

Hi Friends, Do you know if there is a way to integrate this provider with Asterisk for outgoing calls ? Thanks in advance Paulo >>> lists@tlsolutions.net 22-11-2004 23:10:32 >>> Paul, your current method of load balancing is quite fine. Why do you want to round robin load balance? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Paul Hales Sent...

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

...n) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <todd@tlsolutions.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: asterisk-users@lists.digium.com > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [erro...

Hi All, I need some help w/supervised transfer and conference w/a 7940 phone. When I do a blind transfer the calls go through great, but when I do supervised transfer the 7940 tells me "Transfer Denied". When I do a conference call I hit the "conf" key and then dial the next extension. The new call connects and I hit "conf" again but the calls do not get bridged.