With all the talk about servers, how about adding your server hardware
only in a single line without quote to this thread? It will be easier
to tally the results. Mine is a home made PC.
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Jon,
I don't know the purpose of it either but that is what the client wants.
- Original Message -
From: Jon Pounder [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, September 18, 2007 5:59 PM
Subject: Re: [asterisk-users] Interesting Conference Request - Can thisbe
Compaq P3 1GHz server (about 6 or 7 yrs old) running 2gb RAM, 40(?)G HDD,
single AX100P.
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.488 / Virus Database: 269.14.3/1054 - Release Date: 06/10/2007
19:12
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Eric,
I had the same feeling when I found it difficult to find sip hardphones oid
stuff.
Maybe th right thing is to split Users and Ressources data in different
repositories.
Users data might be stored in AD, and Ressources in an homemade database.
I hope that people who disagree wouldn't
Thanks for the info, this is helpful :)
On 10/4/07, Faidon Liambotis [EMAIL PROTECTED] wrote:
Hello,
This is a update on the current status of Asterisk in Debian.
Apologies for the really long mail, it is targetted both to users and
maintainers :)
I'm Ccing asterisk-users as a one-time
Hi List,
I am trying to learn SIP in its entirety. I have so far found:
http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403
Anyone know of any other books that are worth reading ?
Thanks.
Justin
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I have used nufone in the past and the quality is good.
- Original Message -
From: Alejandro Lengua
To: Commercial and Business-Oriented Asterisk Discussion ; Asterisk Users
Mailing List - Non-Commercial Discussion
Sent: Saturday, September 22, 2007 7:48 PM
Subject:
EPIA 5000 wit 512MB RAM, TDM400P (1xFXO) and FritzCard for ISDN. Running
Mandrake 9.2 as later distros I have tried (Fedora) wont play nicely!
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asterisk-users mailing list
To
Home built Jetway VIA CN700 1.2Ghz, Dual GigE, 1G DDR2, Linux From
Scratch 6.3, Asterisk 1.4.12, Samba, LAMP Untangle (shortly)
--
The way out is open!
http://www.theopensourcerer.com
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Alan Lord wrote:
Home built Jetway VIA CN700 1.2Ghz, Dual GigE, 1G DDR2, Linux From
Scratch 6.3, Asterisk 1.4.12, Samba, LAMP Untangle (shortly)
+ Single FXO (x100p)
--
The way out is open!
http://www.theopensourcerer.com
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Hi All,
A couple of weeks ago I noticed Askozia PBX, which is a new embedded
Asterisk OS distro at http://askozia.com/pbx. This caught my
attention for two reasons; it uses v1.4 of Asterisk, and it uses the
m0n0wall development framework to build on FreeBSD with a PHP based
GUI. I've used
Hi Al,
That was it, Thank you!!!
Al lists wrote:
check tz option in your voicemail.conf
On 10/5/07, *Chuck Bunn* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I have a really oddball time problem. When I check the server time
using
'date' it is correct. When I
On Thursday 04 October 2007 23:26:16 Faidon Liambotis wrote:
Backporting stuff from trunk may be error-prone and is not easy to draw
a line of which stuff we should backport.
I'm open to suggestions on other modules that may have sense in
backporting.
There are a number of packages for which
Hi
is there a tool to know what was the maximum calls that asterisk managed?
Thank you
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Il Neofita wrote:
Hi
is there a tool to know what was the maximum calls that asterisk managed?
http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
You can test it with sipp:
http://sipp.sourceforge.net/
Alexandros
Doug Lytle schrieb:
Il Neofita wrote:
Hi
is there a tool to know what was the maximum calls that asterisk managed?
http://areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54
Doug
--
Note: forwarded message attached.
-
Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ---BeginMessage---
Hi:
I installed A102d sangoma's card successfully but Asterisk doesn't answer to
incoming call from pstn and console
Astlinux does seem to be growing cob webs a bit. Askozia doesn't support
Zaptel cards in the GUI and not sure if it is possible to configure them
manually. There is no Voicemail storage mechanism yet. It's still very
basic but a nice start.
-Original Message-
From: Michael Graves
I forgot to add that this is a T1 ISDN PRI line on which I am sending
the DTMF digits.
Regards
Arpit
On 10/5/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hi,
Is there any way to get the DTMF digit preferably in the
extensions.conf . The dtmf digits would be entered by the user
Hey everyone,
On 10/7/07, shadowym [EMAIL PROTECTED] wrote:
Astlinux does seem to be growing cob webs a bit. Askozia doesn't support
Zaptel cards in the GUI and not sure if it is possible to configure them
manually. There is no Voicemail storage mechanism yet. It's still very
basic but a
SIP is only one piece of the puzzle.
This will show you the max calls using SIP but what about TDM? What
about IAX?
I have seen 95 simultaneous calls come in over a PRI with NFAS (Sangoma)
eating 60%-70% of the CPU (HP DL320 3ghz single core with a gig of RAM)
with no codec conversion. Ulaw
Justin Case wrote:
Hi List,
I am trying to learn SIP in its entirety. I have so far found:
http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403
http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403
Anyone know of any other books that are worth reading ?
Hello all,
I have successfully complied and installed 1.4.11 and recently
noticed that 1.4.12 (and subsequently 1.4.12.1). Unfortunately, while
1.4.11 was running fine, 1.4.12.1 seems to build fine, but after a
few minutes, I receive the following message a few times:
ERROR[17670]:
The board never came to market 1. because the demand. 2. impossible
to do with zaptel.
/b
On Oct 7, 2007, at 3:23 PM, Steve Totaro wrote:
How about the once announced Digium DS3 card (that I never saw come to
market), that board must have some powerful onboard circuits or
require
a very
shadowym wrote:
Astlinux does seem to be growing cob webs a bit. Askozia doesn't support
Zaptel cards in the GUI and not sure if it is possible to configure them
manually. There is no Voicemail storage mechanism yet. It's still very
basic but a nice start.
-Original Message-
On Sun, 07 Oct 2007 18:25:36 -0500, Darrick Hartman wrote:
A few comments on this as one of the Astlinux developers. Asterisk 1.4
is and has been in a beta branch for some time. The developers feel
that while 1.4 is the future, in many cases 1.2 is a much more stable
platform. Also while
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten =
Hi List;
From where I can buy the G.729 and G.723 licenses, and
how I can install it on Asterisk so I can use it?
Anyhelp?
Regards
Bilal
Don't let your dream ride pass you by. Make it a reality with
could this be the reason for my problem ?
( I am using a 64 bit AMD processor )
2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED]
* main/stdtime/private.h, main/stdtime/tzfile.h,
include/asterisk/localtime.h, main/stdtime/localtime.c: Working
on issue #10531
On 10/7/07, Justin Case wrote:
Hi List,
I am trying to learn SIP in its entirety. I have so far found:
http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403
Anyone know of any other books that are worth reading ?
http://www.geocities.com/intro_to_multimedia/books.html
Telling someone to read the RFC bah.. might as well give them a
blanket and pillow because they will fall asleep. chan_sip is just
ugly in every way.
/b
On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote:
http://www.faqs.org/rfcs/rfc3261.html
as well as the source in asterisk (1.4.11
Now wait a minute,
I favor books and bookmarks myself, but the young man seems
determined to conquer the kingdom of SIP, so we told him
where the princess is hiding :-)
Maybe he has the fire of a dragon in him and just sipping SIP
wont do.
-baji.
--
On 10/7/07, Brian West wrote:
Hello,
I'm new to this list, but I've been hacking around on Asterisk for a
few months now. I'm getting ready to transition 3 small businesses
that I own to a fully VOIP system with a IAX trunk (no POTS lines)
but I've hit some snags and I'm looking for some good documentation.
I've read
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