I would love to run Asterisk on a BSD system. I do not know of any developers
actively working on Asterisk on a BSD platform, though my knowledge isn't
comprehensive.
It may be worth talking to the people doing the packaging for various BSD
platforms, to see how involved they are, or if they
To follow up the discussion - yeah, it's not RAM, or at least not directly.
I'm so used to looking in the asterisk logs I didn't think to look at
/var/log/messages:
Feb 10 09:10:45 telephone-retsof kernel: [35734.705648] asterisk[11215]:
segfault at ffa2048e ip b70a3def sp b540a000 error 4 in
On 14-02-10 9:46 AM, Mike wrote:
> What log entries are leading you to think that you're running out of RAM?
None. It's just my guess. The log doesn't show anything except Asterisk
restarting.
--
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-- Bandwidth and Colocatio
RAM and have it help, because it's 32-bit. I intend to move to a
64-bit machine, but I was hoping to wait until summer. Does anyone have any
immediate tips for dealing with this sort of rush?
Justin Sherrill - American Rock Salt
P: 585
8* for that one) keeps
going in a loop - downloads updater, saves it, formats the filesystem,
downloads the new bootROM, and then repeats. There's no error on screen and no
successful upload of logs to show an error.
Has anyone updated these models before and seen this?
Justin Sherrill - Ameri
"No Answer" is working fine when enabled. I
was looking at the sip.cfg but don't know exactly what to look for, can you
give me a hint to where would i find that option?
Thanks,
On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill
mailto:justin.sherr...@americanrocksalt.com>>
uot;forward no answer" working?
Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 585-298-6826
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New to Asterisk? Join us for a li
s
like an echo.
I think turning rxgain/txgain down may make a difference, but I haven't tried
it yet. Has anyone else experienced something similar?
Justin Sherrill - American Rock Salt
P: 585-991-6825 F
http://lists.digium.com/pipermail/asterisk-users/2012-February/270427.html
That worked for me with the polycom 3.x firmware; I haven't tried it with 4.0
firmware yet.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David
C
I have used a Plantronics CS351N and a CS70N with Polycom IP550 desk units.
(both are single-ear units, in different forms) Each one needed a Plantronics
APP-5 to replace using a lifter.
They worked fine. The one complaint that I had from users is that the headset
beep to show that a call wa
try using the
extended BLF stuff (described here http://www.excaliburtech.net/archives/147
and here http://www.voip-info.org/wiki/view/Asterisk+presence)
gordu
On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill
mailto:justin.sherr...@americanrocksalt.com>>
wrote:
This is one of those "Is anyone else doi
Out of curiosity, what is "the Polycom script"?
I obviously haven't moved from 3.2.x firmware yet.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 4:45 PM
To: Aster
sh the light next to that
extension, but that's it.
Is anyone using a similar setup and seeing this? It's somewhat rare, but I
have an office location where everyone there likes to pick up other people's
calls, and they haven't been using a call queue like they oughta.
Justin Sh
I've noticed that if I have people on speakerphone at the two farthest
ends of our internal network, they will occasionally get a second or two
of feedback. (sounds like jingle bells) I'm figuring it's some very
slight amount of packet loss or jitter that isn't helped by the
speakerphone echo, ma
Asterisk will send the two SIP endpoints 'reinvite' messages, so that they talk
RTP directly with each other. Depending on your version of Asterisk, setting
the 'canreinvite' or 'directmedia' option may make a difference, since that
will keep the traffic flowing through the servers, and the pho
I've had mystery reboots with Polycom IP550s - the culprit in both cases was
the network connection. Replacing the cat5 cable to the phone or changing the
attached port fixed it both times.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@
Queuemetrics is neat-looking. However, it requires MySQL, and I'm using
Postgres. Does anyone have a recommendation for a different product for
reporting usage that's not tied to MySQL?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lis
Anyone have some recommended equipment for alerting people to calls in a noisy
environment?
I have Polycom IP550 phones set up in some really noisy environments - our mine
hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now.
They're analog devices, attached to Linksy
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On >Behalf Of Carlos Chavez
>Sent: Saturday, January 15, 2011 2:02 AM
>To: Asterisk
>Subject: [asterisk-users] Asterisk stops responding
>
> I am having a problem with
> From: marvin horst [mailto:fivehor...@gmail.com]
> Sent: Tuesday, October 19, 2010 10:23 AM
> To: Justin Sherrill; asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] integrate Intertel Axxess with Asterisk
>
> How did the setup work as far as extensions on th
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On
>Behalf Of Danny Nicholas
>Sent: Wednesday, September 22, 2010 5:04 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Asterisk- speech to text(Voice
I encountered something strange. A local business has an ACD that, when I call
it using a Polycom 550 connected through an Asterisk system, will respond to
button presses only if they are short.
Calling this business with our old (non-Asterisk) phone system or with my cell
phone works because
I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'.
Everything appears normal, but the status of the members never changes from
'not in use', even if they are being rang or are in a call.
Members are added like so:
queue add member SIP/1406 to marketing penalty 0 as SIP/1406
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