similar to: Usage Reports

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

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

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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

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

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

On Tue, Nov 12, 2013 at 5:20 PM, Andrew J. Schorr <aschorr at telemetry-investments.com> wrote: > 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

On 11/14/2013 03:39 AM, Gene Cumm wrote: > On Tue, Nov 12, 2013 at 5:20 PM, Andrew J. Schorr > <aschorr at telemetry-investments.com> wrote: >> 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

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

Hi all, I am trying to install Vicidial in an existent FreePBX installation (I'm using Xorcom packages for Debian Etch), but I didn't find any documentation, I found only this guide [0], but is for trixbox only, do you think it will work on FreePBX on Etch? [0] http://iptn.org/vicidial/index.html Regards, Diego Quintana Cruz

Greetings- First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance. I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,

Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel

Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near

Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Regards Bilal

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

I'm currently reviewing the latest release of FreePBX (formerly known as Asterisk@Home). Do either of you know whether FreePBX is robust enough to handle multiple clients, or have any recommendations on front-end Web interface to manage client config & provide clients access to manage their level of access (similar to how Vonage, Teliax, and others provide client access to their web