search for: vanderdecken

Currently when I call in to my * box it answers after two rings. I'd like for it to answer without ringing. Is this an option somewhere in the dialplan that I'm missing? Thanks, Andrew

I hope someone can help me with this. This is what I want to happen. Someone dials in and goes to my extension. First, the phone on my desk rings If there is not an answer, I would like to have the dialplan call my cell phone. If I answer my cell phone, speak the incomming number to me. I press one of the buttons on my cell phone to accept the call. If I don't answer, or I don't

I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory

Hi - I think I might have seen this problem on the list before, so I'm sorry if this is a duplicate, but I couldn't find it when searching through the archive.... I'm just setting up a new machine with asterisk. It's a RH9 box, and I've tried the RC2 tarball, the 1.0 CVS and the 1.0 RPM's from nacs.net (thanks). My config is basically the sample barebones sip setup

...hink redhat makes so much on support contracts? Personally, I'm only not a newb cause I paid for a training class to get me out newbieism(sp?) My 2 cents -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Race Vanderdecken Sent: Wednesday, March 02, 2005 8:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Why should I answer a Newbie question,therethick! If some one would like to show me the site that explains how to setup a mailing list then I will create a Ne...

Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering

Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office

Hey all again, I have successfully compiled and am running Asterisk (stable release) on OS X (10.3). However, any make directive that uses the "-shared" option in gcc results in an error. Apple states that -shared is not supported under OS X. Is there a workaround or do I have just have to live life without those modules (zaptel, libpri, format_mp3, probably others)? Thanks, Zach

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.

> -----Original Message----- > From: Iassen Hristov [mailto:ih.ng@databrokers.net] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?

...hat allows only G.729, then use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Thu, 17 May 2007 11:22:17 -0400 > From: "Race Vanderdecken" <asteriskusers@codetyrant.com> > Subject: RE: [asterisk-users] cpu usage for G.729 codec > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <01d101c79897$2c802df0$0e01a8c0@Pre...

...hat allows only G.729, then use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Thu, 17 May 2007 11:22:17 -0400 > From: "Race Vanderdecken" <asteriskusers@codetyrant.com> > Subject: RE: [asterisk-users] cpu usage for G.729 codec > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <01d101c79897$2c802df0$0e01a8c0@Pre...

Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible

Hello there, I am playing around with Asterisk the first time and it really looks great. ;-) However, I have one problem: Any SIP device can connect to my PBX. How can I requre logon for SIP users and deny access in the case of wrong or missing credentials? Thanks Florian