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Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution dahdi-linux-2.1.0.4 I am getting following error *echo "You do not appear to have the sources for the

I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch G?es

...efforts of many people involved. That done, maybe we would conclude an interactive script would be the missing piece to incorporate user choices that are hard to default to. > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.cohen at xorcom.com<jabber%3Atzafrir.cohen at xorcom.com> > +972-50-7952406 mailto:tzafrir.cohen at xorcom.com > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >...

Hi, I am running Fedora 7 VM. On an earlier configuration with zaptel and Asterisk 1.4.21 , meetme worked alright. I upgraded to Dahdi and Asterisk 1.4.26, and the result is choppy sound via Meeme, while a simple Musiconhold works OK with descent audio quality. So I am sure its a Dahdi_dummy problem. Running dahdi_test gave me very poor results. Opened pseudo dahdi interface, measuring

Hello, Is there any app_backticks (see http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or workaround for 1.6 ? In the past, I had trouble trying to use ENV() function. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090407/966917a0/attachment.htm

Hi, Today I'm building Asterisk using steps like this : http://mikeoverip.wordpress.com/2008/03/29/asterisk-compilation-and-installation-on-debian-etch/ As you can see, the first requirement is to download various dependencies such as gcc, g++. As I'm trying to centralize everything (configuration files, source codes in an SVN repository), I'm wondering if there is a smarter way to

Hi, It seems BRI signalling settings are missing from http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf I would like to add those parameters : bri_cpe_ptmp bri_cpe bri_net Is this http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conftied to a specific Asterisk version ? Can I edit this ? Regards -------------- next part -------------- An HTML attachment was

Hi, When reading logs, I can see a couple of lines such as : full.6:[Jun 30 15:53:26] NOTICE[6599] chan_dahdi.c: PRI got event: Alarm (4) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6599] chan_dahdi.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 full.6:[Jun 30 15:53:32] NOTICE[6607] chan_dahdi.c: Alarm cleared on channel 1 full.6:[Jun 30 15:53:32]

Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card ???? Thanks a lot Alejandro

Is there a way to restrict connection to my asterisk server to users based on their IP addresses, and not just password. I have some hackers who connect to my server to make illegitimate solicitation calls to people. I had to shutdown the server for now until I find a solution. ANY HELP? Thanks. ond -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, using Asterisk 1.8 beta3 installed from scratch I am not able to stop/start/restart Asterisk deamon with /etc/init.d/asterisk stop|start|restart It just happens nothing, no warnings, errors etc. I am running Debian Lenny. Any ideas what is wrong? Thanks, Oliver -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,

Hi, Has anyone information about BRI hardware supported by 1.6 libpri ? In another thread, I was told a basic BRI card with HFC chipset (Bewan Gazel 128) was supported but I would delighted to lear about other harwarde (and specifically about Digium B410P). Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hy guys i am having so much hard time to setup asterisk on a virtual machine that i got , i just want to know if i really need to use Dahdi and libpri on a complete Digital PBX i just gonna use sip and iax. I will never use any kind of analog line on this machine. Wait for a feed back. Daniel Abreu.

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

After a reboot, I can't reproduce the problem anymore which is quite frustating. 2011/6/14 Tzafrir Cohen <tzafrir.cohen at xorcom.com> > On Tue, Jun 14, 2011 at 03:44:32PM +0200, Olivier wrote: > > Hi, > > > > I'm using a two-years old installation script for the first time on a > > Squeeze (linux 2.6.32) platform. > > For an unknown reason (might

After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports

hello: I wan to use the test tools-patgen and pattest for pri cards. according to Tzafrir Cohen from http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know how to use that. do i need to connect two pri cards with two servers, or use a cable to connect two cards in one server? please give me a more details in term of cables and configurations. thanks! Chris -------------- next part

Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the " == Parsing '/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have