similar to: ztdummy configuration help

Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe("H323:996", "") in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno'

I have ztdummy installed: Module Size Used by ztdummy 3464 0 zaptel 218756 1 ztdummy crc_ccitt 2176 1 zaptel ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 piix 8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 rtc

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. cat meetme.conf [rooms] conf => 600 extensions.conf: [meetme] exten => 2663,1,MeetMe(,D) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set

Hello System: Debian etch with kernel 2.6.18-4-686 or 2.6.18 custom. Asterisk Version: SVN-branch-1.4-r55483M Zaptel Version: SVN-branch-1.4-r2302 modules all ok in compilation time. And modules loaded: ztdummy 5928 0 rtc 13364 1 ztdummy zaptel 181540 1 ztdummy crc_ccitt 3200 1 zaptel In /dev/zap directory I have:

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a

Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] --

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten =>

Hello, I ran into this problem today, I am trying to implement "One click installation", I followed foreman howtos and set up the pre-requisites accordingly. However, when I click on "Build" button, I get the following errors in the foreman''s console and another error in the web interface indicating that the installation failed. Any ideas? *"PuppetCA: SSL/CA or

Hello, I have enabled summary emails from foreman and set up a cron job which sends me periodic summary emails. However, the mail''s content is sort of plain text. Now this is more like a feature request rather than a problem, I am wondering if we could use some kind of HTML template and pass the values through this HTML template and then mail the output as a summary email to the

Hi all, I have a problem starting the ztdummy. Here is what happens: [root@asterisk /]# modprobe ztdummy 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 ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing

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

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

Hey Guy! I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten =>

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

Hi list, we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686 kernel. The server has 2 B410P cards plugged in. No other card. We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with chan_misdn, all is fine. In misdn-init.conf we have added option=1,master_clock. Asterisk is up and

>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() >If I assign the DID to ring extension SIP/2000 and then after time-out send >it to MeetMe() or Playback() it works and the caller

Good afternoon, I'm trying to use MeetMe in an AGI script written in Perl, as follows: print "EXEC MeetMe 2000|p \n"; $res = checkresult(); The problem that I have is that when a user press '#' in order to exit from the conference, everybody goes out. This is randomized because sometimes doesn't happened. My current version of asterisk is: Asterisk

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

Hi, thanks for your mail! On Sat, 17 Jul 2004, Chris Lattner wrote: > > That is wonderful! Wow, you did this just ~1 month? :) Yes :), even less, but that is since I used the structure from SICP, see the URL below. > Cool, ok. Have you seen the LLVM GC support that is already available: > http://llvm.cs.uiuc.edu/docs/GarbageCollection.html > > It should be able to support