Displaying 20 results from an estimated 20000 matches similar to: "No subject"
2006 Dec 15
2
Bandwidth requirements for 1, 000, 000 minutes a month
This may expose my ignorance, but here goes :)
I've been asked to figure out how much bandwidth would be needed to handle
1,000,000 minutes a month.
Here's the environment:
) All calls are received via SIP.
) All calls use the ulaw codec.
) Calls average 10 minutes in duration.
) The "busiest" hour will account for 10% of the daily total.
This is how I'm figuring
2009 Jul 20
0
No subject
expected context is valid (may not work on 1.2, I started this ride at 1.4
and therefore have no backward knowledge).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2014 Mar 21
3
Need more meetme users -- hitting some limit
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
CentOS 6.5.
The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
The primary application will be bridging groups of users using meetme().
I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1
behaving a bit more like a production box -- bridging calls (box2).
The call
2005 Aug 23
2
SIP powercycle not hanging up
I have Sipura 841's talking to a CVS-HEAD of August 4.
If I disconnect the power to the Sipura, Asterisk does not hang up the
channel.
My sip.conf for this phone looks like:
;
[super1] ; Sipura 841
disallow = all
allow = ulaw
callerid = "super1"
2009 Aug 29
0
asterisk-users Digest, Vol 61, Issue 84
On Sat, Aug 29, 2009 at 10:30 PM,
<asterisk-users-request at lists.digium.com>wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help'
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your =
recipient is using a codec that isn't ulaw or alaw).
=20
_____ =20
From: asterisk-users-bounces at lists.digium.com =
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel =
freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jul 20
0
No subject
might be your best bet to get the information you want. I'd look at
voip-info.org for information.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 16, 2009 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to list ongoing calls
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
2007 Apr 09
1
Re: asterisk-users Digest, Vol 33, Issue 35
We i have settup it like this it giveme caller id agent id and date-time on
gsm file but i want them to be in folder on every day basis datewise.
exten =>
_1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP})
exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)
Any Idea ?
Faisal
> ------------------------------
>
> Message: 16
>
2015 Jun 26
2
Asterisk 13 logging to two places
On Fri, 26 Jun 2015, Dale Noll wrote:
> I added a filter to the /etc/rsyslog.conf file
>
> :syslogtag, contains, "asterisk" stop
>
> Syslog is still receiving the messages, but is discarding them.
Nice to learn a new (to me) feature of rsyslog.
What does 'logger show channels' show?
--
Thanks in advance,
2009 Jan 16
0
No subject
FYI, not sure if it's of use to you... but... The digium tc400b is a transcoder card that can offload upto 120 channels of transcoding for g729 <-> ulaw... It's available as PCI only, but, if that's OK, it could be an alternative to replacing your server... G729 licenses are not needed when using that card...
There have been posts by some people about having multiple CPU
2011 Sep 02
0
No subject
built-in; This doesn=92t matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web =
interface.
Let=92s say Yves=92 =93special conference=94 is 5555. The moderator =
would start
using this command
Exten =3D> s,1,meetme(5555)
The participants would do
Exten =3D>
2015 Apr 27
2
adding area code
On Mon, 27 Apr 2015, Bryant Zimmerman wrote:
> exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1})
Missing a colon?
${EXTEN:-1}
--
Thanks in advance,
-------------------------------------------------------------------------
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Newline Fax: +1-760-731-3000