similar to: conference problem without zapata interface

Hello all ! How can I make conference authorization based on pin number ? I have: exten => 1,1,Meetme,1234|ps|2222 where 2222 is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke

Hi, It is possible to create a dummy account (SIP or IAX type) in order to be used in a "dummy" extension? I want to be able to use it as a normal extension (as an IP phone connected to it), but without the need to answer or call from that extension. I want that when I call that extension to hear the ring, and after the defined period of time to enter in the Voicemail system. I

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:

hi, i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? thanks a lot for the help!!! Angelo

Has anyone solved the problem on the ADSI phones that when you hit one of the soft keys, the Number Pad stops working?

I've seen this happen a few times and I think it's when the system that Asterisk is running on crashes due to a power failure (or for some other reason that causes a non-planned shutdown). While Linux comes up fine, Asterisk won't start because the drivers are loading in the wrong order. fixed by: 1) sh /usr/src/fix-asterisk-modules.sh 2) sh /etc/init.d/asterisk start Is this a

Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the

I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is working fine. However, I want the asterisk.adsi to load into the 'self-load' slot but can't figure out what the correct FDN for doing this is. Does anyone know the right FDN for the SL slot on these phones? Also, does anyone have any cool/interesting ADSI scripts they wouldn't mind sharing? I'm

I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing

In this message I try to summarize what I have learned in these last two weeks. My primary sources of informations were the * list archives and linux ISDN docs. I ain't no * master, so don't trust too hard. Relevant messages from the * list for the current discussion are: 009177.html 009268.html 0498.html 0849.html My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2)

Hi All, I recently purchased the Asterisk Developer's Kit (TDM) to try out Asterisk. After following the directions in the Digium's FAQ topic entitled "Q. How do I configure my TDM40B and X100P?", I'm receiving the following error: WARNING[1074428608]: File chan_zap.c, Line 6748 (load_module): Ignoring rxwink ERROR[1074428608]: File chan_zap.c, Line 6692 (load_module):

Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

I would us Meetme for conferance SIP-->SIP fist. my Meetme.conf: [rooms] conf => 9999 my extensions.conf: exten => 9999,1,MeetMe(9999) But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable

Hi All, I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P (numbered 5-8). Everything works fine except that I cannot use one of the FXS ports (number 5). If I configure zapata.conf to recognize it, the whole system voice quality suffers. I've tried already to switch PCI slots, with no results. Below is a snapshot of my /proc/interrupts, maybe this can shed some light on

Hello, I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. Something like this: (THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)