similar to: Incoming caller presentation doesn't work - out of ideas

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Easy questions for you guys probably, I'd like to serve 10 parallell incoming calls at the same time, so I bought a lot of Zap-channel cards for analog phone lines. But I want all users to be able to use the same phone number to dial in, but I want the number to be switched to an avaiable zap-channel. Do I need some kind of switch for this? It sounds reasonable, but I'm not sure. :) Am

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

Hi, I'm using SIP MESSAGE to asterisk V10 and it fails to be received. I'm not super sure of the reason but I'm making this guess: Due to I'm using non ipaddress in the to field, which contains sip:mobil1.xyz.com, Asterisk makes the mistake to try matching this name "mobil1.testserver.com" in extensions.conf and no extension/peer is found in the sip-message context

We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice,

I'm running a small Linux server with samba installed on it I want to access this server from an XP client which is a member of a ad-domain but when trying to logon XP putts in ad-domain-name\username as logon name to the samba server, how can I work around this?

Hello group, I am installing Asterisk on a new pc running Ubuntu 12.04.3 Server. ?I have dahdi-complete-2.7.0.1+2.7.0.1, and have tried 2.6.1 and 2.8.0-rc1 and 2.8.0-rc2. ?I am following along in the ASTERISK The Definitive Guide 4th Edition and on page 116 when I issue the command $ sudo /etc/init.d/dahdi start I get the following error: Loading DAHDI hardware modules: ? ?wctdm24xxp: done

I just installed a TE820 octal span T1 card, and it's not showing up in dahdi_hardware output. This was installed into a test machine that already has a TDM800P card in it, and that one is showing up and working fine. Is there some kernel module that I'm missing? Lspci: 05:04.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) 21:08.0 Communication

Hi All, I have a server running Fedora 14, kernel 2.6.31.14, Asterisk 1.6.2.17.2 and Dahdi 2.4.1. I have the wctdm24xxp+ loaded with a Wildcard TDM800P with 8 FXO ports When a call is placed extension to extension, there is no problem... When an extension is used to dial out, the called number answers, but the server only detects the answer about 3 seconds later .. The outgoing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I live in Sweden and I am having problems getting asterisk to properly detect when a caller hangs up. And yes, I DO have disconnect-supervision on my line. Also asterisk sometimes misinterprets the disconnect-signal as another incoming call. This usually happens if I hang up first and then when the caller hangs up, asterisk treats it as a new

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not .....

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130309/d82ddd47/attachment.htm>

Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very

Hi All, This is a request for testing for users of Asterisk 1.4 or 1.6 with any of the following Digium VoiceBus based cards: TDM2400P, AEX2400, AEX800, TDM800P, TDM410, TE120P, TE121, and/or TE122. From a practical standpoint, this branch should allow these boards to work in more systems / configurations where IRQ misses caused problems in the past. The updated wctdm24xxp and wcte12xp

The Asterisk.org development team has announced the release of Zaptel versions 1.2.20 and 1.4.5. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler

The Asterisk.org development team has announced the release of Zaptel versions 1.2.20 and 1.4.5. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler