>> marek cervenka <[EMAIL PROTECTED]> writes:
>>
>>> hi,
>>>
>>> i'm testing asterisk 1.4/1.2 in the following scenario
>>> centos5/cpu quad xeon E5335 2.0Ghz
>>> - test clients behind nat
>>> - 1500+ testing instances - reregister option from 1min to 1hour
>>> - qualify set to 5000
>>>
>>> top sho
> marek cervenka <[EMAIL PROTECTED]> writes:
>
>> hi,
>>
>> i'm testing asterisk 1.4/1.2 in the following scenario
>> centos5/cpu quad xeon E5335 2.0Ghz
>> - test clients behind nat
>> - 1500+ testing instances - reregister option from 1min to 1hour
>> - qualify set to 5000
>>
>> top shows over 100
On the polycom manual it says for g729 use rfc1890. I did that but
sometimes it works sometimes it doesn't. Am not sure why.
Jarga Jallow
Technical Support Engineer
2985 S. Hwy. 360
Grand Praire, Texas 75052
Direct: 972-206-1212 ext# 29
Mobile: 214-669-9046
Fax:972-999-4113
Toll Free: 1-877-80
You need to do a 'make' before the 'make install'.
Lyle
[EMAIL PROTECTED] wrote:
>
> Hi all,
>
> Please help me in installing Asterisk.
>
> I am getting the following error when trying to install Libpri
>
>
> [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2
> [EMAIL PROTECTED] libpri-1.4.2]$ make cle
Sip.conf : ; Note: If your SIP devices are behind a NAT and your
Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on
[EMAIL PROTECTED] wrote:
>
> Hi all,
>
> Please help me in installing Asterisk.
>
> I am getting the following error when trying to install Libpri
>
>
> Please help me out.
>
> Thanking you,
>
> Preeta Pandey
You aren't compiling the latest version of 1.4.3. Have you tried that?
If that
Hi all,
Please help me in installing Asterisk.
I am getting the following error when trying to install Libpri
[EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2
[EMAIL PROTECTED] libpri-1.4.2]$ make clean
rm -f *.o *.so *.lo *.so.1 *.so.1.0
rm -f testprilib libpri.a libpri.so.1.0
rm -f pritest pridu
Hi,
I don't know spa3000, but in spa-3102, to transfer call to PSTN to asterisk i
change the following, i'm new in asterisk but i think it's correct:
PSTN tab,
DIAL PLAN 8: S0<:[EMAIL PROTECTED]:5060 (ip of asterisk)
Proxy: 192.168.2.2:5060
outbound proxy: left blank
sip udp: 5061
subscriber
I will be out of the office starting 01/24/2008 and will not return until
01/25/2008.
I will be in the Billerica Data Center and will be checking email
periodically. I will make every effort to respond to emails in a timely
fashion and emergency requests should be directed to the IntelliCare CS
I have been using Dell servers and have no issues with linux, in fact when i
implemented my last install with their top of the line server (dual xeon
quad core and SAS drives on Perc 6i) i was amazed how smoothly it went
trough.
Beside that i like their open manage, it runs nice on linux and its a
What I did was to change the dial plan to ring exttension at the
asterisk ip address -- which under freepbx simulates an incoming call -- this
is the only way
I have ever gotten that function to work properly.
What does not work for me is voip-to-pstn -- I get a 403 response from
the spa3102.
Hi, all
I am trying to figure out how to forward incoming PSTN call on SPA3000
to VoIP extension(s).
Basically, I have converted my home to VoIP. I have normal phone
(connected to SPA3000) and couple of IP phones. All call coming from
VoIP DID do ring all phones (analogue via SPA3000 and IP ones)
Hi
On Jan 25, 2008 4:58 AM, John Faubion <[EMAIL PROTECTED]> wrote:
> I have the same issue but I haven't put much effort into solving it yet. Too
> many other issues seem to get in the way.
>
If you do, please post your results !
___
-- Bandwidth and
On Thu, 2008-01-24 at 16:28 -0500, Ken Mink wrote:
> I can not seem to get the two to talk together. I am running asterisk
> 1.2.21.1. I am seeing the following repeatedly in *
> Jan 24 16:23:40 NOTICE[17063]: chan_sip.c:11291 handle_request_register:
> Registration from '2345 ' failed for '10.15
Ronald Wiplinger wrote:
> Can anybody give me a hint, please.
>
> I have a Welltech FXO device and from PSTN coming calls will be
> transfered to the extension number 1001.
> I want that the caller can reach the extension number by dialing
> said number.
>
> My 1st try was:
>
> exten => 1001,1
I have been given a Patton SmartNode 4114 and asked to get it working as
POPS gateway for our asterisk box. The 4114 has 4 FXO ports. It's got
firmware 3.21 on it. I currently have a single POPS line plugged into
port 0.
I can not seem to get the two to talk together. I am running asterisk
1.2
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Andrea Cristofanini wrote:
> yes i see
> you have to enable in ADVANCED SETTING
> Challenge Response on Phone: = OFF
> Regards
Sweet as, cheers man.
- --
Kind Regards,
Matt Riddell
Director
___
http://www
My polycoms all have dtmfmode=rfc2833 and they work fine on both
asterisk's IVRs and external ones brought to me from the PSTN:
[120]
type=friend
context=internalaugmented
secret=a_secret
host=dynamic
*dtmfmode=rfc2833*
Moj
Jarga Jallow wrote:
>
> Hi,
>
> I am having trouble making a selection
Please post your sip.conf entry for your phone and also describe your
calling path. Are you having a problem with internal calls (e.g.: to
voicemailmain) on the same switch, or are you referring to calls to
PSTN destinations via pots/pri/sip/? Also, which versions of
Asterisk, Zaptel, linux, etc.
On Thu, 2008-01-24 at 13:51 -0500, Steve Prior wrote:
> Matthew Rubenstein wrote:
> > Is anyone else interested in creating new voices for Festival (the
> > voice synth bundled with Asterisk) that might not be as good as
> > Allison's recordings, but are better than the current Festival voices?
Allison just replied to me and said she recorded new 'brighter' sound
clips. I have 1.2.1 sounds, do I need another version to have these new
sounds?
On Jan 24, 2008 11:56 AM, Erik Anderson <[EMAIL PROTECTED]> wrote:
> On Jan 24, 2008 10:14 AM, Matt <[EMAIL PROTECTED]> wrote:
> > That worked...
Matthew Rubenstein wrote:
> Is anyone else interested in creating new voices for Festival (the
> voice synth bundled with Asterisk) that might not be as good as
> Allison's recordings, but are better than the current Festival voices?
If you try to do live voice synth for prompts you'll proba
Hi,
I am having trouble making a selection when I call a number and need to
make a selection to go to an extension with my polycom phones 301.
Anybody have an idea how to fix this problem?
Thanks in advance.
Jarga Jallow
Technical Support Engineer
2985 S. Hwy. 360
Grand Praire, Texas 7
Is anyone else interested in creating new voices for Festival (the
voice synth bundled with Asterisk) that might not be as good as
Allison's recordings, but are better than the current Festival voices?
On Thu, 2008-01-24 at 12:00 -0600,
[EMAIL PROTECTED] wrote:
> Date: Thu, 24 Jan 2008 11
> No one else is seeing this issue ?
I have the same issue but I haven't put much effort into solving it yet. Too
many other issues seem to get in the way.
John
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Huh?
As far as I can see, the latest 1.2-release is 1.2.26.1, at least in
tar-balls...
Hmm.. OK, looking in svn I now see the 1.2.26.2 release... Shouldn't it be
in tar-balls as well?
Another hmmm Interesting... After looking at the download page again
now, about 5 minutes after my lastest
its kinda tangential to the original post,
but check this link
http://www.voip-info.org/wiki/index.php?page=Asterisk%20Native%20Sounds
since the link for download doesn't seem to work I can post or send
the files in many codecs for you
On Jan 24, 2008, at 9:56 AM, Erik Anderson wrote:
> On Jan
On Jan 24, 2008 10:14 AM, Matt <[EMAIL PROTECTED]> wrote:
> That worked... hrmm not that great... anyone know of any decent sounding
> recording of Allison for Asterisk?
What's your definition of "decent sounding"? IMHO and that of many of
my co-workers, the default Allison recordings sound great.
Hi
On Jan 21, 2008 11:05 PM, Jean-Yves Avenard <[EMAIL PROTECTED]> wrote:
> This works great. However in the CDR, than seeing one entry for each
> call, I see several entries in the CDR
> Worse, if I do something like:
> Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED])
That worked... hrmm not that great... anyone know of any decent sounding
recording of Allison for Asterisk?
On Jan 23, 2008 11:26 PM, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
> for x in *.g711u; do mv "$x" "${x%.g711u}.ulaw"; done
>
> On Jan 23, 2008 5:00 PM, Matt <[EMAIL PROTECTED]> wrote:
>
On Thursday 24 January 2008 08:54:03 Jaswinder Singh wrote:
> Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve
> SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic'
>
> Quite obvious .. doest sippeers have that row ?
Or download 1.2.26.2.
--
Tilghma
Yes, it does... This is not a problem of usage. This mail should
probably have been sent to -dev instead, as it clearly is a bug.
If you take a look at the link I provided you will see that this is
indeed a bug, and it has been fixed in 1.4 branch in 1.4.17. The problem
is that 1.2 is in "secur
On 1/24/08, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
> Hi, all. I've done some Asterisk recelling, but recently got roped into a
> Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much
> all circuit-based systems do, it sucks. It sucks to administer, moves
> suck... you know th
That's funny. I seem to remember installed Deb/* on a Poweredge 2950.
.must be slowly losing my mind. Another side effect of using Asterisk?
Dementia?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gopal krishnan
Sent: Thursday, January 24, 2008 2:51 AM
To: [EMAIL PROTECTED]
Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic'
Quite obvious .. doest sippeers have that row ?
On Jan 24, 2008 6:04 PM, Torbjörn Abrahamsson <
[EMAIL PROTECTED]> wrote:
> Developers and maintaine
I like the Echo application in asterisk ;) . Weird :P
On Jan 24, 2008 7:07 PM, Mark Johnson <[EMAIL PROTECTED]> wrote:
> Ken D'Ambrosio wrote:
> > Hi, all. I've done some Asterisk recelling, but recently got roped into
> a
> > Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty m
Ken D'Ambrosio wrote:
> Hi, all. I've done some Asterisk recelling, but recently got roped into a
> Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much
> all circuit-based systems do, it sucks. It sucks to administer, moves
> suck... you know the drill. So, I'd love change to
yes i see
you have to enable in ADVANCED SETTING
Challenge Response on Phone: = OFF
Regards
/a
Matt Riddell ha scritto:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Has anyone ever seen an Snom320 lose settings?
>
> It's been working fine for months and then I got a call this morn
Developers and maintainers, any information?
// T
Torbjörn Abrahamsson wrote:
> Hello!
>
> We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some
> problems when using realtime for peers. We connect the PBX to a sip peer
> at an ITSP, and when we try to dial the peer we get:
>
I recently went through the same thing. My company was paying huge amounts
of money for voice PRIs at several locations, ongoing PBX support to a third
party, and huge amounts of money for a teleconference bridge to yet another
third party. I was bored one weekend so I implemented Asterisk. Before
Does anyone remember this site? http://www.astertest.com/
Regards,
Ariel
On Wed, 2008-01-23 at 18:30 -0500, Steve Totaro wrote:
> Link?
>
> Thanks,
> Steve Totaro
>
> On Jan 23, 2008 6:08 PM, Paul Hales <[EMAIL PROTECTED]> wrote:
> >
> > There was a cool paper written a a few months ago where
marek cervenka <[EMAIL PROTECTED]> writes:
> hi,
>
> i'm testing asterisk 1.4/1.2 in the following scenario
> centos5/cpu quad xeon E5335 2.0Ghz
> - test clients behind nat
> - 1500+ testing instances - reregister option from 1min to 1hour
> - qualify set to 5000
>
> top shows over 100% cpu. cpu c
Hello,
On Jan/24/2008, Paul Hales wrote:
>
> http://www.transnexus.com/White%
> 20Papers/asterisk_V1-4-11_performance.htm
>
> It was the bottom news item on voip-info.org - I was worried I would
> have to really search for it!
and I guess that transcoding benchmark could increase to non-transc
On Wed, 23 Jan 2008, Tilghman Lesher wrote:
> On Wednesday 23 January 2008 23:23:23 Anthony Francis wrote:
>> Tilghman Lesher wrote:
>>> On Wednesday 23 January 2008 12:23:24 Gordon Henderson wrote:
Is there any way to find-out the peak number of calls that an asterisk
system has had? No
On Thu, Jan 24, 2008 at 01:20:59PM +0530, Gopal krishnan wrote:
> Hi,
>
> Dell is not a recomeded server for linux. Its only compatible with
> windows.
And I suppose you have checked this. And specifically paid a short visit
to Dell's site before.
http://linux.dell.com
http://linux.dell.com/di
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