similar to: Not able to compile asterisk, zaptel, libpri in /usr/src

We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem to actually do anything. Is there some secret/trick to getting a report out of it that will tell us which extensions are placing calls? I've tried every query on the form that I can think of. Is the reporting disabled by default or ??? Any tips/pointers appreciated. Ben M. Schorr Chief Executive Officer

We deployed a single phone handset (Polycom 331) at a remote site. We have a IPSEC VPN running between the firewall at the remote site and the firewall at the site where our Asterisk/FreePBX box lives. We have used a similar configuration for this site before and it worked fine. We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the

How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows "(ULAW)" (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be

O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium TE205P connected to a single span if ISDN PRI. The Telco has assigned us two local numbers to test incoming calls. I created an inbound route for one of those DID's and assigned it to one of our extensions. Sounds simple enough. Too simple, apparently, when I dial the number the caller gets a recording that it's a

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, On Tue, Nov 12, 2013 at 01:41:29PM -0800, H. Peter Anvin wrote: > On 11/12/2013 01:22 PM, Andrew J. Schorr wrote: > > I am using 4.05 because that's what's packaged in the latest Fedora > > release (Fedora 19 in my case). I wonder why they are using such an > > old version. I will take a look at how difficult it would be to build > > the latest version on

hi friends, i have installed asterisk,libpri,dahdi tar files in /usr/src. problem is that when i compile (./configure) asterisk-1.4.26.3, gtk+2.0.0 dependency is missing. i installed gtk from gtk.org, now when i am compiling gtk (./configure), i am getting this error message configure: WARNING: *** TIFF plug-in will not be built (TIFF library not found) *** configure: error: *** Checks for TIFF

Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -----Original Message----- From: asterisk at opensourcesolution.in [mailto:asterisk at opensourcesolution.in]

i have installed database POSTGRESQL for storing call details. when i restart database i get the error. [root at localhost server]# psql -h 127.0.0.1 -U asterisk Password psql: could not connect to server: Connection refused Is the server running on host "127.0.0.1" and accepting TCP/IP connections on port 5432? THIS IS MY /VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF # CONNECTIONS AND

hi friends, as i am a beginner in voip, i had made a very simple dial plan i had made two extentions n both are able to ring each other through soft phone (X-Lite) below is my dialplan ################################################################################################################################### /ETC/EXTENSIONS.CONF [others] [my-phones] exten =>

As I said, please keep discussion on list. asterisk at opensourcesolution.in wrote: > hi all, > > first of all i appologise for sending u pvt email. i have installed > asterisk on Centos 5.3, plz open the attachment in which i had drawn a > tolpology. i had installed one asterisk machine and two windows machine. > now i want to install softphone in both windows machine.

hi friends, i had installed postgres database for call log,call detail. it has restarted succesfully but when i check tcp connection i dont get any welcome message by psql. [root at localhost ~]# # psql -h 127.0.0.1 -U asterisk password [root at localhost ~]# th in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of

hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever

This is why I don't do this kind of work anymore. Impossible to distinguish the phantom problems from the real ones - and I'm convinced there ARE phantom problems when you install new telephones on people's desks. Suggestion: learn to use the facility in Wireshark that can log a SIP/RTP stream and report actual latency and packet delivery stats. That will give you some solid info

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --

Hi, Thanks for your help. On Mon, Nov 11, 2013 at 08:00:47PM -0800, H. Peter Anvin wrote: > There is no reason by Syslinux would reboot the system on a serial > disconnect. There could be a bug, of course; now, Syslinux 4.05 is > pretty old so unless you can reproduce this on a current version that > would be useful. I am using 4.05 because that's what's packaged in the

plz do not send for me e-mail thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091015/d76bfa3d/attachment.htm