similar to: Zap timing device

Hi, I have this error, I have a digium card TE110P Tiger3xx When I'm load the dirvers by this command modprobe wcte11xp I got this error "Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for wct1xxp " when I'm Lunch the command ztcfg -vv I got this error.

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 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

I've got an * system that is having some real problems with 1.0.2. The biggest problem is that calls going through my T100P get choppy for about 10 seconds every 1 or 2 minutes. Asterisk is running on a debian stable system with current packages. The T100P is plugged into a Adit Channelbank with 8 POTS lines hooked up to the Channelbank. I've watched the vritual memory and CPU status on

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

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

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

Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar

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Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. My musiconhold.conf is ; Music on hold class definitions ; [classes] [moh_files] default = >/var/lib/asterisk/moh-native ;default =>

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 all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there

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