similar to: asterisk meetme confrance problem

Dear all I have video confranceing deivice Codian and i want to intergrate asterisk box with codian so voice confrance is possible with codian users means some users have not codian endpoint so thay call join confranceing with SIP PHONE I have configures asterisk and register codian in asterisk now whn i call from asterisk to codian i got IVR and ask me to inter confrance

Hi all, Our domain is running perfectly, but for the fun of it, I tried a samba-tool dbcheck, and (unexpected actually!) it returned many errors, on many users, like this: > ERROR: wrongly formatted userParameters on CN=username1,CN=Users,DC=samba,DC=company,DC=com, should not be psudo-UTF8 encoded > Not changing userParameters from UTF8 encoding on

I was hoping someone might have some knowledge to impart regarding VOIP PRI Gateways or the psudo ISDN services being offered these days. The official line in Australia is that "true" ISDN services are on their way out. I am testing a service provided by one of the telcos I am told that it cannot provide ISDN cause codes for disconnected/invalid numbers and all we get is the audio. Also

Hi, I recently upgraded 4.1.8 installation to 4.4.2. As per documentation I ran samba-tool dbcheck. It reported some problems with userParameters: ERROR: wrongly formatted userParameters on CN=,OU=Users,OU=MyBusiness,DC=,DC=local, should not be psudo-UTF8 encoded. I ran --fix as I read in some discussion that this was actually tested. The problem after upgrade is, that it is no longer possible

Hi all, Appologies for the rather basic IO question but I am rather new to R... Migrated from IDL/Matlab recently. I have a rather simple Fortran control file (sigh...) that I am trying to parse and read using R. My problem is that the file's format is somewhat flexible. Imagine: --- 1> 39 1901 2> 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 3> 22 24 26 28 30 32 34 36

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hi, when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and start asterisk to be able to use MeetMe(). When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and start asterisk but I am not able to use MeetMe(). What do I miss? Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

Hi Peoples I am having problems with meetme, in that it responds with "conf-invalid" when I dial a conference number. I notice that there is a note with regards to ztdummy, and the need for that to be loaded. Is this still the case? Is meetme dependent on this module? I do NOT use zaptel cards in my system, and there for zaptel is not loading. Can anyone shed some light

Hello all, I am planning to setup a MeetMe conference functionality on Asterisk-1.4.13without having a Zaptel card. All users will be calling through SIP only. AFAIK, the said application needs a timer which makes use of the ztdummy module. I have basically two (2) problems I am encountering here that [1] I can't load the ztdummy.ko module and [2] Asterisk don't run when running it

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

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

On Tue, 2015-12-01 at 12:16 +0100, samba list wrote: > Hi all, > > Our domain is running perfectly, but for the fun of it, I tried a > samba-tool dbcheck, and (unexpected actually!) it returned many > errors, > on many users, like this: > > > ERROR: wrongly formatted userParameters on > > CN=username1,CN=Users,DC=samba,DC=company,DC=com, should not be > >

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: "That is not a valid conference number." On the console I get: "unable to open pseudo channel". As

On Thu, 2004-01-15 at 19:18, dkwok@iware.com.au wrote: >> I do not have any zaptel hardware on the Asterisk box, I could not have >> meetme functioning. I did modify the Makefile in zaptel directory on >> line 168 by including ztdummy as one of the modules to compile in. try modprobe ztdummy This works. Should I include this in /etc/asterisk/modules.conf so that it will

We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave

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

I'm running into a bit of a problem setting up conference calls. The box I rent at a colo doesn't seem to have USB hardware.... When I try to load usb-uhci I receive a "device does not exist" error. Which means I can't load ztdummy.... The system has a rtc clock module, so zaprtc won't work... (which I'm scared to unload rtc because I don't have physical access

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>