similar to: what interface for ISDN-10/20/30?

Hi All, I know this is not specifically Asterisk related but I don't knew where else to ask for help. Does anyone know how to or if it is even possible to allocate 512kbit/s to an ISDN device from a 30B+D ISDN line. The building the office is in has a E1 30 channel service (30B+D) but we can not get any 2B+D ISDN services. I have a HDX Polycom video conferencing system that requires a

What's the difference between the TE121 and TE122. I read the description on Digium's site and it isn't clear to me. Best regards, -- Guilherme Loch G?es Visite nossa loja virtual: http://www.shopvoip.com.br Not?cias e F?rum sobre VoIP com software livre: http://www.asteriskexperts.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't

Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, I'm having a problem with CallerID names not showing up when calls come in. I have dialplan code to store the callerid(name) away and it is blank (null). However, the voicemail variable ${VM_CALLERID} has the name field populated. For example, here is some of the dialplan code: 2. Set(CALLER_ID_INFO_ALL=${CALLERID(all)}) 3. Set(CALLER_ID_INFO_NAME=${CALLERID(name)}) 4.

Dear asterisk users, We want setup TE121 digium board: Model: Digium TE121: VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface. http://www.digium.com/en/products/digital/te121.php My platform Server: HP Proliant 150 G5 OS: UBUNTU x86_64 GNU/Linux Asterisk: 1.4.21.2 zaptel: SVN-branch-1.4-r4662M When we enable zaptel driver for TE121, the

Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root at Asterisk12 ~]# asterisk -rvvv asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call

Hi fellow asterisk users, I am running Asterisk 1.4.29 with an Digium TE121 card (wcte12xp kernel module) an approx. 100 snom320. The whole installation is running without issues since 5 months. Without having changed anything on the asterisk server for at least 2 months, i noticed "clicking sounds" at external calls over the PRI. Asterisk constantly throws the following messages: ...

Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: ? checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci:0000:04:08.0 wcte12xp- d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA

Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius

Hi, I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The setup consists of the following components: - A Digium TE121 for connectiong to E1 ISDN - Debian box with Asterisk 1.4 - Grandstream GXW-4008 SIP ATA to which the Fax machines connect I am aware of the problems with this type of ISDN <-> Asterisk <-> SIP ATA <-> Fax machine installations, e.G.