similar to: TDM box Hardware

Hello, I?m trying to compile the Fritz CAPI module for Debian stable, following the steps related in http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install But I always get the same error, debian-asterisk:/home/ismaelg/fritz# make (cd src.drv; make CARD=fcpci) make[1]: Entering directory `/home/ismaelg/fritz/src.drv' cc -c -DMODULE

Hello, I trying to configure asterisk to store sip and iax2 user in a mysql database. All goes well, but my problem is when i try to add a new user (sip or iax). I have look for an aplication with a web interface that lets us manage the user account in asterisk without success . How could I manage the users without making sql query throw the mysqlclient? Do you know any application that

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension 1111 picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a "w" in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w5555555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz card to do the telephony. Will there be less feature if doing so? -- David Kwok, CISSP Tel: 612

Has anyone had any echo issues with the Digium TDM02B FXO card? I purchased a clone wildcard on E-Bay for $8.00 and have had horrible echo issues.... I'm assuming it is because of impedance issues? Just wondered what people's takes were on the TDM02B FXO and echo?

Greetings to you all, I had a quick question that I was not able to find an answer for. I am running Redhat 7.2 on i386 (Pentium III), and I am getting an error when I try to compile the samba 2.2.2 source, and I get the following error: Using FLAGS = -O -Iinclude -I./include -I./ubiqx -I./smbwrapper -D_LARGEFILE64_SOURCE -D_FILE_OFFSET_BITS=64 -D_GNU_SOURCE

Hello, I am trying a couple of days before to set up asterisk to redirects an incoming call if the extension dialed is busy without success. I just try to use 'Gotoif' command, with bad luck, it can't do what i want. Anybody could helpme? ani clue will be appreciated. Regards. Ismael.

Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I previosly defined. Someone could help me? Any clue will be appreciated. Regards from Spain. Ismael

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

Hello all, I'm trying to setup a AVM C2 card. I have read the kernel requirements for this card. <M> CAPI2.0 support [*] Verbose reason code reporting (Kernel size +=7K) [*] CAPI2.0 Middleware support (EXPERIMENTAL) <M> CAPI2.0 /dev/capi support [*] CAPI2.0 filesystem support <M> CAPI2.0 capidrv interface support My problem is when I make a "make

Hi all, I have done a major upgrade, first I have bought a Digium TDM400P (TDM02B) 2-Port FXO to replace x100p cards, than moved from an old version Asterisk to a new centos 4 box, compiled asterisk from Asterisk CVS-HEAD built on 2005-06-11. Seems that my iaxy is now "incompatible" with the TDM400P. When I receive a call from the TDM02B, the caller can't hear my voice (I am using

Gilberto Nunes via samba wrote: > Oh Gosh! > > Why blame me just because I expose my option about samething?!? > I have work for years, since 1998, if I remmember with Linux, Samba, > Iptables, Mysql e other tools.... > And I will continue to work with these tools and never stop.. > I already try OmniOS, Oracle, RedHat ( since RedHat 5 ), Slackware.... > I am not a noobie!