similar to: Single ringer phone for incoming calls, that every one can answer

Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the

it work with just the main ringing phone and not have it pickup the rest of the phones in your group. That's more what you are looking for. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mitcheloc Sent: Saturday, June 23, 2007 10:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Hi, I am trying to install the Asterisk-addons-1.4.2, and when I make install it prompt me such error messages make[1]: Entering directory `/usr/src/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory

Hi, Does Asterisk_addons_1.4.2 cant be use in the new Asterisk release or no one really want to share their experiences? Since this project is belonging to everyone within this list, why still no one really want to share the experiences and to growth the Asterisk to the next level by keeping their secret in behind. As see, Asterisk 1.4 has so many feature improvements, and it's

Hi all, Just want to check from the list experienced personal about the Digium HPEC, where I had purchased the HPEC and wish to run with TDM card Sangoma A200. I can't install HPEC to run with Sangama A200 card, even I had changed my hpec file from i686 to i386. The error that I had as bellow; > Found key 'HPEC-XXXXXXXXXX' for 2 channels. > Found valid HPEC licenses for 2

Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike -------------- next part -------------- An HTML attachment was

I noticed that the BudgeTone (I have the 102) with the latest firmware tries to download a file called cfg.txt (presumably the configuration) and a file called ring.bin (presumably a ringer) from the tftp server. The ring-in sound on the budgetone is the same as the ring-out sound and that is going to be confusing for users. I contacted GrandStream and was informed that both of these formats are

Asterisk 1.2.13 I am trying to figure out the best way for a night bell at work. Note: I have no spare buttons available on the phones. But I do have two lines and two park positions as buttons. Option 1 (easiest and the one I just implemented) When asterisk is in night mode, Connect to IVR, List all options and then if they dial 0 or timeout, ring every phone in the

Tom Lanyon <tom at netspot.com.au> wrote: > On 06/12/2007, at 8:20 AM, James A. Peltier wrote: >> > Tom Lanyon wrote: >> >>> >> Hi all, >>> >> Trying to install 5.0 or 5.1 (i386 or x86_64) on a new system from >>> >> an install DVD. As soon as the installer launches anaconda, the >>> >> screen shuts off

Hi all, New to the list and (mostly) new to CentOS. We have just moved from running RHEL to CentOS on our main production servers. Anyway, I am configuring a couple of new SunFire v20z servers here at the moment and trying to get the systems' management interfaces to work fully under CentOS. First issue was trying to install the openipmi RPM from the CD that came with the SunFires, this RPM

Hello, I'm new to the list but I've been following it since its inception. First of all sorry for the length of this post, if you're not a Fedora user just skip it. I've been building Compiz packages since Fedora Core 5 came out (Xgl only) and updating since then with all the new stuff, AIGLX and whatever. I use them on a very large range of hardware. Attached are my source rpm

Il 02/08/20 19:09, Alessandro Baggi ha scritto: > > Il 02/08/20 18:54, Stephen John Smoogen ha scritto: >> On a side note, you keep emphasizing you aren't expecting an SLA.. but >> all your questions are what someone asks to have in a defined SLA. I >> have done the same thing in the past when things have gone badly, but >> couching it in 'I am not asking'

Hi list, The bug reporting guide in the wiki mentions we should notify the - devel list when reporting a new bug, so I just wanted to bring some attention to: http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1328 I''d reported another bug myself (http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1332 ) and marked it as a duplicate of the one above as they seem to be the

Hi list, Trying to upgrade someone's workstation here to 5.2 (was installed from a 5.0 DVD I think). The RPMs on our internal mirror are in-tact and pass a 'rpm -- checksig' test, yet when I run a 'yum upgrade' a large portion of them are corrupted and fail the GPG check. This seems to be isolated to yum, as downloading the RPM directly via FTP with wget or lftpget

If SLA supports IP trunks, can shared extension appearance be achieved using a local SIP trunk in place of an extension? Basically, I'm trying to allow some stations (Polycom IP 650) to have a shared extension amongst all of them. Ideally, I'd like for the LED to show if that extension is in use, and I'd like for the extension to ring all stations on that extension when a call comes

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

anyone using/experimenting with this new feature in asterisk 1.4? is anybody able to post some info how to use and what features are supported? I have general knowledge how SLA should work, ie. monitor status of another line like BLF with additional features like answer ringing call, barge into existing call on shared line and make conference call or resume call, that was put on hold on

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

I am using SugarCRM together with the asterisk plugin, which allows me to click a number, SugarCRM calls my extension then places the call when I pickup. I would like to have that extension auto-answer. I set it up as line 3 on my phone so normal calls do not get auto-answered. However, I have not been able to get this to work. Has anyone implented this? This is what I put in the config file

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw