search for: vanderdecken

Displaying 20 results from an estimated 34 matches for "vanderdecken".

2004 Sep 30
4
Setting auto-attendant to answer immediately
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
2005 May 20
3
Help with follow me
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
2005 Jan 12
2
Trouble building appradius
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
2004 Sep 24
1
No sound into asterisk???
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
2005 Mar 03
4
[OT] - Why should I answer a Newbie question, therethick!
...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...
2005 Mar 08
4
force SIP authentication
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
2005 May 31
3
Opinions of Sphinx?
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
2005 Jun 09
5
Voicemail and MS Exchange
> -----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
2005 Feb 20
2
How many line appearance can Snom 200 handle?
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
2005 Feb 25
1
Asterisk and 723,729
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
2005 Feb 25
1
SIP Errors
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
2005 Mar 21
2
Compiling with gcc -shared on OS X
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
2005 May 20
1
How can you keep agents logged in across a restart?
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.
2005 Jun 10
1
Re: Voicemail and MS Exchange Synchronizatio n
> -----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
2005 Jun 13
2
Need Help with pickup *8
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
2005 Sep 09
2
"Registered SIP '202' ... expires 1800". Why does it expire
Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?
2007 May 18
0
Re: asterisk-users Digest, Vol 34, Issue 82
...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...
2007 May 18
0
cpu usage for G.729 codec
...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...
2005 May 30
1
asterisk compatible, hot swappable PRI card
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
2005 Jan 11
3
requiring logon for SIP users
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