As the topic states.
In a 20 phone PBX enviroment. I am wondering which would
have the greater server and router load.
What do you use ?
Thank you.
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] Best bet ... IAX vs SIP
Well,
I think you are asking the wrong question here, I think the proper
question would be:
In a 20 extension iPBX environment, what combination of signaling and
codec would provide the
best performace on a hardware of [specify your hardware here]?
Nir S
mr. barker
me at any time to discuss
or business. Constructive criticism is always welcomed.
Thank you all for business and we look for more in the future!
Garrett Smith
VoIPSupply.com
[EMAIL PROTECTED]
716-250-3408 Direct
mr. barker wrote:
I tried calling their toll free number and toll number last
I tried calling their toll free number and toll number last
week in the morning and afternoon and was handed a recording saying this number
is no longer in service. The web site was up but there was no message on
the site as to why the phone numbers were not working.
I just called the
I would like to be able to have multiple users (the wife and
kids) to be able to access the Broadvoice account at the same. No
complaining that way from them J.
I seen someones configuration in the group here but now I
cant find it (lost my glasses). If someone could post theirss or
the
Subject: [Asterisk-Users]
Connecting 2 * Together-Pulling hair out
I have read the docs on connecting 2* together but am unsure
of a few things
Do I need a different account for each number that will be
called from one box to the other ? ie. Do I set up a user account
in call routing on the trunks to forward the
call to the right place. Are you using AMP or trying to do it manually.
I found everything a little confusing as well, but it is simple now that I
understand it.
Chris
- Original Message -
From: mr. barker [EMAIL PROTECTED]
To: 'Asterisk Users
=from-internal
Regards,
Chris
- Original Message -
From: mr. barker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 1:58 PM
Subject: RE: [Asterisk-Users] Connecting 2 * Together
I have read the docs on connecting 2* together but am unsure
of a few things
Do I need a different account for each number that will be
called from one box to the other ? ie. Do I set up a user account on one and
then have the other box log into that account when it whats to make a call
Has anyone tried to install Hylafax on Centos ?
If so is there an rpm .. or what was your compiling
procedure ?
Thanks in return
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Hardware Pentium 1.4 Gig 1 Meg ram 1
FXO100 Card Sipura 2000 Local Network Router SMC Codec 711
Asterisk @ home (lastest)
On average it take almost 10 13 Secs to make an
outbound call to a local number.
Is this a normal time ? Is there something that can be done
to cut this time
The question is in the
logical route that asterisk takes when reading and executing the scripts.
Please see the (?) questions beside the lines.
The goal is not to comment
the lines exten = snip in the [ext-local] everytime that I make
a change using the AMP GUI. Also it would be nice to
You can turn off the amount of logging in the log.conf setting. As far as
the registration goes .. that would be under your Sipura Settings.
You may only want to reduce this to 60 sec registration .. I find that any
longer sometime effects longevity of server to find you in the route.
Only my
(SIP/something,15,t)
exten = 1,6,StopMonitor
exten = 1,7,Voicemail(u804)
exten = 1,8,Hangup
exten = 1,102,StopMonitor
exten = 1,103,VoiceMail(b804)
exten = 1,104,Hangup
Kyle
mr. barker wrote:
I am sure that this was answered somewhere but my lack of being able
to find an answer using google I
Here it is
exten = s,1,answer
exten = s,2,SetCIDName('PMG')
In a lot of config files I see exten = s,snip ..
Is s just an extension or system variable for all extensions ? or
something else ?
Thanks
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I am using [EMAIL PROTECTED]
When I manually add anything to the
extensions_additional.conf file it gets rewritten when I add an extension using
the web interface
I am trying to include the monitor function .. I got that
working however it gets deleted when I add something using the
extensions_custom.conf because
as you noticed all of the AMP contexts have an include
like include = ext-local-custom to link them to
extensions_custom.conf
--- Robert Webb [EMAIL PROTECTED] wrote:
On Tue, 12 Apr 2005 14:05:06 -0500
mr. barker [EMAIL PROTECTED] wrote:
I am using [EMAIL PROTECTED
I am sure that this was answered somewhere but my lack of
being able to find an answer using google I turn to the pros.
What would be the easist way to record all conversations
using Monitor command with the latest [EMAIL PROTECTED] ?
Using a FXO card with SIP extensions
I have
I am using the latest AMP with [EMAIL PROTECTED] I used
the AMP upgrade script
My problem is this I am able to record a greeting
using *77 and it put the recording into the
/var/lib/asterisk/sounds location with the
extentsion preceding the recording
However it seems that
Receptionist Recorded Greeting LocationProblem
Where did you find the upgrade
script? I have not been able to find this.
THanks!
Wiley
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mr. barker
Sent: Tuesday, April 05, 2005 9:25
AM
To:
asterisk-users@lists.digium.com
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