similar to: FW: * and Cisco routers

I am completely new to * ( I know read the archives but this is a little different case) I am trying to setup a Sip system out side my security firewalls for home users. I currently run a Cisco avvid solution internally but it's highly firwalled. I am planning on building a pri out of my 3745 cisco router and pluging it in to a 3810 which is on the outside setup with sip and running a *

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 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

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Does anyone have any recomendations for a free Windows softphone, SIP or IAX that supports the following features: * Message Waiting Indicator * Consultative Transfers * Speed Dials -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040518/ed93a1b4/attachment.htm

Bonjour, En utilisant le logiciel Samba je me suis rendue compte qu'il est possible de supprimer n'importe quel dossier ou fichier, qu'il soit ou non prot?g? contre l'?criture, ce qui pose un ?norme probl?me de s?curit?. Avez-vous connaissance de ce probl?me ? S'agit-t'il d'un simple probl?me de param?trage ? Dans le cas d'un bug, avez-vous une solution ? nous

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

Ok assuming I don't want to record calls using * but instead want a dedicated server that listens to a mirror port and records calls. Is there a cheap software package out there for doing this for mgcp/sccp? I know if evern cut over to * there is a way but I doubt I will even cut 100% over to * so I was wonder what the list has heard of for call recording via sniffing my gates. I know there

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

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:

Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >

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

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

I would like to either talk to over the phone a * admin a few question on the deploy and a few question on the setup of * at the very least if some doesn't mind or a walk thru of there site if there in the Houston area. I am looking at cutting over to * for some things but I have some setup question that could easily be answer in about 30-60 minutes. Please reply to lists efastfunding com

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

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

That 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:

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.

On Wed, 23 Jul 2003, Yifang Dai wrote: > I wish! My company just spend a lot $$ on the shinny CCM phone system, so I > don't think I can change that easily... But if I can get asterisk to > talk to CCM via h323, and prove it's usefulness, I might have a chance > to use * in the branches... Well, good luck, then! > By the way, do you know if we can get *'s VM to

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