similar to: ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere

im getting: Fatal error: Unknown function: mssql_get_last_message() in /var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on line 415 to the wiki.. Jason

Hi all, I would like to know if there is a solution to this question. Scenario: Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these

Dear list, I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). Can someone tell me if the timing device is needed for voicemail and other applications too?. I am

Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242

Hi, I a new user of asterisk, I'm trying to in install zaptel drivers on my ISDN card Digium Tiger 3xx TE110P. And my configuration is # # Zaptel Configuration File # span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = it ; ; ; Zapata Configuration file ; [channels] immediate=no switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming usecallerid=yes group=1 channel

hello, im having trouble with asterisk with medium load, it seems im running out of files, here is a chunk of the logs with grep "\(file\|pipe\)": Apr 18 15:40:46 WARNING[11644] res_agi.c: Unable to create toast pipe: Too many open files Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 WARNING[11574] channel.c: Channel

I got the same problem yesterday and i just ch'ange the kernel from 2.6.12-1.1372_FC3 to 2.6.9 kernel of fedora core 3, and now it work perfectly. I think there is some problems with the new kernel of fedora. -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Altus Snyman Inviato: mercoled? 27 luglio 2005

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Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. ######################################################################## #### # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16

Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone

We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil. What versions does everyone use without problems. 0.59r is PERFECT bkw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040506/937ae19c/attachment.htm

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Hi, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI:

Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from

Hi, I've just got a TE110P card and installed at Asterisk. I configured zapata.conf, according to www.digium.com/index.php?menu=configuration, but the following error is happening: ... ... ... [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara

I have two X100P card in my box. I want to connect regular phone (not the phone line!) to one of thse cards. Does anybody think about the same? I don't really want an expensive solution buying additional card with FXS port, I prefer to make something by myself. It'll be great if somebody can point me to technical materials or show electric scheme of such converter. I believe it should