Displaying 20 results from an estimated 900 matches similar to: "dovecot crashing?"
2009 Jul 20
0
No subject
asterisk -rx 'core show channels' | grep DAHDI | sort -n
Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
----- "Mike" <list at virtutel.ca> wrote:
>
>
Hi,
I have just recently been using DAHDI, and I wanted to know how to
2009 Jan 16
0
No subject
AGI is executable.
=20
Then run 'agi debug' from the asterisk cli, place a call and see what
was send and receive from your agi
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] AGI PHP script
=20
I have the
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
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 Jan 10
0
No subject
Class: default
File: /var/lib/asterisk/moh//reno_project-system
File: /var/lib/asterisk/moh//macroform-robot_dity
File: /var/lib/asterisk/moh//manolo_camp-morning_coffee
File: /var/lib/asterisk/moh//macroform-cold_day
File: /var/lib/asterisk/moh//macroform-the_simplicity
Class: none
File: /var/lib/asterisk/moh/.nomusic_reserved/silence
2007 Jul 12
0
No subject
=20
Thanks!=20
=20
=20
Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
=20
------=_NextPart_000_003E_01C8C00B.B3A8DA60
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2015 Apr 14
2
Re: VM Performance using KVM Vs. VMware ESXi
On 4/14/2015 4:02 PM, Dominique Ramaekers wrote:
>
> Dear Jatin,
>
> Maybe it’s a good idea first to implement Spice:
>
> <video>
>
> <model type='qxl' ram='65536' vram='65536' heads='1'/>
>
> <address type='pci' domain='0x0000' bus='0x00' slot='0x02'
> function='0x0'/>
>
2009 Jul 20
0
No subject
message.
The user hears the recording being played, begins to leave a message and is
disconnected about 10 seconds into the call.<o:p></o:p></span></p>
<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p> </o:p></span>=
</p>
<p class=3DMsoNormal><span
2007 Jul 12
0
No subject
Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
=20
------=_NextPart_000_0452_01C8BF32.9F7C4290
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2007 Jul 12
0
No subject
"We have created an easy and cost effective way to have customized
recordings done quickly and with no hassle."
I thought this was rather amusing, as:
1. If you want multiple prompts recorded, you need to submit a new order for
each, which means that even prompts of a couple of words are still charged
at $12. That is NOT cost effective. You could record all your prompts as a
single
2007 Jun 15
0
No subject
using Asterisk.
=20
Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?
=20
As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to
2009 Jan 16
0
No subject
"What is CentOS?
CentOS is an Enterprise Linux distribution based on the freely available
<ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat
Enterprise Linux. Each CentOS version is supported for 7 years (by means of
security updates). A new CentOS version is released every 2 years and each
CentOS version is regularly updated (every 6 months) to support newer
2009 Jan 16
0
No subject
Telco, location, ect?)
At X times of day?
=20
Ect, ect.
=20
It sounds like bleed over, which can be causes by some many things the
best place to start is to find a pattern if there is one.
=20
James Shigley
Monroe Telephone Answering Service
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC
Sent: Tuesday, May
2007 Jul 12
0
No subject
What is the problem with SIP retransmits?
-----------------------------------------
Sometimes you get messages in the console like these:
- "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet."
- "retrans_pkt: Cancelling retransmit of OPTIONs"
The SIP protocol is based on requests and replies. Both sides send
requests and wait for replies.
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run
Apple doesn't accept (for the moment) an application runs in the background=
. So, when Siphon doesn't run, the SIP server of your provider doesn't know=
your iPhone."
--_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_
Content-Type: text/html; charset="us-ascii"
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>
2004 May 04
2
Can Asterisk support R2 signaling
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
>From: asterisk-users-request@lists.digium.com
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs
>Date: Tue, 04 May 2004 13:32:00 -0500
>
>Send
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