similar to: advice re: Page() application

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.

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

Hi list, I have a customer which will organize a conference in a big meeting room which has a sound system. He would like to connect this sound system to a MeetMe room so participant in the MeetMe can act as if they where on site. My idea is to take a barbone or Notebook, connect it to the sound system using the soundcard and run a softphone on it. Does some of you already have success in

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

Hello all, I'm in final testing stages and preparing training for a new Asterisk rollout. I'm replacing a Cisco Call Manager system, and re-flashing the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, I use the "Confrn" softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and

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

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

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

> Thank you all for valuable input, > > another question: when do I actually need the echo cancellation > (hardware / on board /on module ) ? > It depends on your environment. If there are still analog devices in addition to VoIP, I'd say always, but Asterisk has a rudimentary echo canceller already on board. The Telcos use echo cancellers themselves, but it cannot hurt to

Hi there I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not hear their voice half a second later (because meetme mixes the voice and sends to everyone in the conference). Now because of requirements

Here's a strange thing. I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For conference rooms we're using Polycom IP6000's. We bought two of them brand new. When I configure one phone with a username(SPIDR-3758)/password , it works fine. The other phone won't register with it's user(SPIDR-3749)/pass pair. When I try to use the first phone with the second