similar to: Ast12 issue "missing" library file??

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

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.

>On Fri, Oct 18, 2013 at 03:16:08PM -0400, Cassius Smith wrote: > 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

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

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 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 installed Asterisk on a new Server, it is a Dell Server and has 4 Ethernet ports. I gave IP address 192.168.0.3 for one Ethernet port. I am able to login for asterisk using /usr/sbin/asterisk -rvvv and from there (in the command line) I can type a commands. I have an Polycom IP Phone that is able to register for other Asterisk boxes (and some of them is 1.8.3) but with this new

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

Hi!I've a problem using ssh without password: I want use rsync for automatic scripts,I'm using this 2 names for my asterisk@home2.5 linux (based on red hat), rsync11 and rsync12. This is the way I use to change the configuration and then using without password , but the password is always asked: [rsync11@asterisk11]$ ssh-keygen -t rsa Generating public/private rsa key pair. Enter file

I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call