similar to: Polycom 331 freezes connecting to FreePBX

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

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

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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

I have 2 Asterisk servers connected via IAX2 connections. PBX1 is on the internet with a public IP Address - with PRI PBX 2 is behind a NAT router with IAX2 Ports forwarded 1-way audio is an issue with incoming and outgoing calls using the PRI. However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there are no issues. As well as a restart of asterisk on PBX2

Thanks, Thomas. This machine is running RHEL enterprise. I installed with RedHat Package Manager. -B -- *Bjarne Bartlett* | Graduate Student | Biostatistics & Quantitative Health Sciences | 651 Ilalo Street, Honolulu, HI 96813 | (610)864-7435 | bjarne at hawaii.edu *WARNING*: E-mail sent over the Internet is not secure. Information sent by e-mail may not remain confidential.

Aloha Paul, I did originally try to view the server on port 8000 and 8001 and I get an error message that the site cannot be reached. It initially seemed like a firewall issue, but our LAN admin has opened the ports and allowed the icecast connection. Interestingly, when I start icecast it hangs and does not print that the server has started, is this an issue that anyone has encountered? I have

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On 11/11/2013 05:38 PM, Andrew J. Schorr wrote: > Hi, > > I hope somebody can help me with this problem. I don't know whether > it's a problem with my configuration or a bug or feature in extlinux. > I am using an IPMI 2.0 serial-over-lan console to connect to a system > with an Intel S3200SHL motherboard. The first line of my extlinux.conf > file looks like this:

On 11/12/2013 01:22 PM, Andrew J. Schorr wrote: > 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

On 24 Jul 2019, at 21:37, Bjarne Bartlett wrote: > Aloha Paul, > > I did originally try to view the server on port 8000 and 8001 and I get an > error message that the site cannot be reached. It initially seemed like a > firewall issue, but our LAN admin has opened the ports and allowed the > icecast connection. Interestingly, when I start icecast it hangs and does > not print