similar to: Asterisk devel. - Mediatrix dtmf bug solved

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Hi, We are currently testing Asterisk with Wildcard 100XP and serveral Cisco ATA Box. Everything works great except that the card does not detect the hangup signal. We are using a standard Belgian PSTN line. I have not found anything about a be zone (only us, fr, de, nl, ...). Does someone experience the same problem? Do I need to create a new zone be (and how to do that)? Another small

Hi, We are using Asterisk running on FreeBSD, as IVR / Voicemail for SER. We have redirected certain calls from SER to *. On * there is some 'testing' extension. It's simply playing some demo now ;-) As long as I use plain G.711 the sound is nice. When I switch to G.729 the sound is choppy, not recognizable. What is going on? Debug shows everything is normal.. I understand that all

> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in > routing calls to upstream carrier via SIP trunks out.? I spent a lot of time > in the lab testing 1.8 which included heavily testing DTMF with no issues > that came up.? It all just seemed to work fine.? But then again you can?t > reproduce every real work scenario in the lab. > > > > I?m

I've had a few problems with my system holding the line after a call has been made, just now I rebooted and noticed the following in /var/log/messages Aug 13 17:23:15 Sheriff kernel: wcfxo: DAA mode is 'FCC' in the file wcfxo.c the following structure is initialised as below which would suggest that FCC is wrong for France or pretty well all of Europe. static struct fxo_mode {

Hi, I'm currently making some configuration with Asterisk and I wonder is someone has already a sample settings of indications.conf for Belgium. Thanks. Emmanuel Bergmans ---------------------------------------------------------- Perceval Technologies sa/nv Rue Tenbosch, 9 B-1000 Brussel Tel: +32-2-6409194 Fax: +32-2-6403154 General:info@perceval.net -

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

[sorry for the loss of proper attributions, this is from two messages]: [Me] >This is something I've encountered in trying to make a particular > asterisk application handle properly IAX2 frames which contain either > 20ms of 40ms of speex data. For a CBR case, where the bitrate is > known, this is fairly easy to do, especially if the frames _do_ always > end on byte

Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations?

Jean-Marc Valin wrote: >>Heh. I guess after playing with different jitter buffers long enough, >>I've realized that there's always situations that you haven't properly >>accounted for when designing one. >> >> > >For example? :-) > > I have a bunch of examples listed on the wiki page where I had written initial specifications:

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

adding gsm or just comment out the disallow and the 2 allows. (your = recipient is using a codec that isn't ulaw or alaw). =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi list! I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend of mine. Just an * server and for outbound calls wengo.fr was used to place calls via sip. He had a strange echo on the line I didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of

Hi all I want to look at setting up a simple / cheap SAN / NAS server using normal PIV motherboard, 2GB (or even more) RAM, Core 2 Duo CPU (probably a Intel 6700 / 6750 / 6800) & some SATA HDD's (4 or 6x 320GB - 750GB). My budget is limited, so I can't afford a pre-built NAS device. Can this be done with CentOS? I've been looking FreeNAS (which is built on FreeBSD), and it

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: