search for: zapta

Hi, I would like to use the asterisk box with zapta card to enable some conferencing. I would like to use only TDM connections without VoIP. I'd like also use the Meetme app. I have some questions: 1. Does any one use it for a few conference rooms at ones ? 2. Is it possible to restrict the number of users connected to one conference room ?...

Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's poss...

...xfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in zapta.conf, but never get to a point of receiving anything more then the above at the cli. The fax machine is an older Brother unit. Any thoughts? Rich

...1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please explain how to do this in detailed. I want to know how to route all outgoing calls through our SIP server and how to stop some of the extensions from taking outgoing calls Thank you, Kanishka -------------- next part -------------- An HTML attachment was sc...

My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message "the person you are calling, is currently unavailable" This is an ubuntu machine, with a sangoma card, with FreePBX setup, on asterisk 1.4. Incoming calls are working fine - outgoing national, mobile, a...

...n Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1 port FXO card). So, do you have to have mulitple Zapata devices to enable multiple users in a conference? Do I have to enable the zdummy (dummy zapata code) to run a multiuser conference? Thanks Lee Goodman PS, if someone could send a...

...sk in the following way 1>FXO connected to PSTN line 2>the calls coming to PSTN line should be received 3>SPI clients should be able to call outside through PSTN 4>There is no phone connected to the FXS In this case , do i need to bother about configuring FXS , in zaptel.cof and zapta.conf provided I am not going to connect a phone to FXS Thanks Joseph John ___________________________________________________________ Win a BlackBerry device from O2 with Yahoo!. Enter now. http://www.yahoo.co.uk/blackberry

Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4...

...'ve been able to set them up so I can dial extensions "123" and "124" to talk between them. I'm able to access the default "1000", "500", and "600" extensions and they all seem to work. Now my problem starts to occur when I installed the Zaptal T100P card into the system and compiled the kernel and driver (Zaptal 1.0 driver). The driver module appears to load fine when the system boots up. Once asterisk starts and I use one of the X-lite phones to call "1000", "500" or "600" they appear to connect, but I...

...t added a line to the config. >Now using the number it gave me, in the trunk config in AMP, I still cannot >get an outside line (connected it to a simple analogue pbx >system) and call outside the *-server.. >Could anyone help me with this? >Thanks guys" You need to go into the Zapta.conf and remove the semi colon ; channel => 1 Jeff

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

...all until 2 rings after the existing Nortel PBX sees it. Both people calling in and people answering don't like that. I've gone through whatever documentation and mailing list archives, but haven't been able to find working solutions. Have tried various combinations in zaptel.conf and zapta.conf but no luck yet :-( Ideas anyone? Thanks, Wim

Can I have a TDM400 and a T100P in the same machine? I am using AAH and trying to combine two boxes. If so, can anyone tell me the proper config for zaptel.conf and zapata.conf? Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051006/061d9d84/attachment.htm

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I'm using app_festival to speak some text to callers. I'm having two problems with this. The first is with IAX calls (I've not tried others) the first few seconds of the speech is garbled. The second problem I'm having is the the volume of the speech IS VERY LOUD. I tried putting the following in the siteinit.scm but it didn't seem to make any difference. (set!

...answering of Call waiting calls? And how do you answer call waiting calls with the system. I have usecallerid=yes, hidecallerid=no, callwaiting=yes, callwaitingcallerid=yes, threewaycalling=yes, transfer=yes, cancallforward=yes, callreturn=yes, echocalcel=yes and echocancelwhenbridged=yes in the zapta.conf file. How do you use these features? Or where can I find some documentation. I found some information about *0 on the thread but I couldn't find the corresponding config information. Also, is thier a list of functions like *0 and how to use them. Also, on another note. Using this set...

...ivacy manager/Zapateller not working correctly. I have posted the details of this problem in a previous post, but basically, they don't seem to recognize anonymous callers vs callers with no caller-id. 3) Call-waiting caller-id doesn't work. (but either by themselves do) I am not using zapta interfaces, but I have enabled the appropriate setting in that conf file in case it has some relevance. I didn't see a similar option in the sip.conf file. I have tested with both the ATA186 and the SPA-2000. 4) I am looking for a provider of caller-id name database services on a U.S. natio...

...heir documentation that they do have impednace matching. Did Carrier Access make a typo, or does it depend on the version of the FXO card? I current have a CAC AB1, and I am having echo problems with analog and sip phones. I have echocancel=128, echocancelwhenbridged=128, echotraining=yes, in zapta.conf. 128 taps for echocancel does the best job for my system... I've adjusted the rx gain and the tx gain to the lowest possible, and stilll be usable. I've ran a brand new cat5 from the dmark. And I still have echo problems on incoming calls only; outgoing calls are perfect. Would a...

...placement will come in a week. In order to make all perfect at once, I rather ask one more time before doing it. I have installed Asterisk before on another machine, which I will use now for this week. I copied the /etc/asterisk directory from the last running machine to this one. I disabled zapta.conf, since I am not panning for this few days to plug in the Digium card. 1. problem # modprobe ztdummy modprobe: Can't open dependencies file /lib/modules/2.4.21-273-smp4G/modules.dep (No such file or directory) Why, and how to fix it? 2. I have started to use realtime and have the hard...

Hi i have a big problems with my asterisk .. i use a Digium TDM400P for connect a analog line. And not all time (i don't know why) he don't see the end of the call and anyone can call me (occuped) For that's work, i am disconnect the phone cable and it's good anyone have a idea ? bye