Displaying 20 results from an estimated 2000 matches similar to: "No subject"
2007 Jul 12
0
No subject
to use the table asterisk.cdr but I can't find it anywhere.
------_=_NextPart_001_01C92A7E.C6B88024
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" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
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 Jul 20
0
No subject
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Error loading module
'chan_dahdi.so': libpri.so.1.4: cannot open shared object file: No such
file or directory
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Module
'chan_dahdi.so' could not be loaded.
=20
I am using on CentOS 5.4 64 bit.
Asterisk 1.6.0.25
Asterisk-addons
2009 Jul 20
0
No subject
=20
arp | grep "192.168.0.1"
=20
substituting the IP address of the SIP device.
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
=20
hello,
is
2013 Aug 27
0
[LLVMdev] unsubscribe
<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=Content-Type content="text/html; charset=gb2312">
2009 Jul 20
0
No subject
=20
arp | grep "192.168.0.1"
=20
substituting the IP address of the SIP device.
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
=20
hello,
is
2008 Nov 16
0
No subject
===========
process 5496: arguments to dbus_connection_ref() were incorrect, assertion
"connection != NULL" failed in file dbus-connection.c line 2497.
This is normally a bug in some application using the D-Bus library.
org.freedesktop.DBus.Error.Spawn.ExecFailed: dbus-launch failed to
autolaunch D-Bus session: PuTTY X11 proxy: wrong authentication protocol
attemptedAutolaunch error:
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 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 Jun 26
0
number of samples in input_frame
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
That is a dumb question :)<br>
<br>
The encoder expects SPEEX_GET_FRAME_SIZE at all times. If you are
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
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
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 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
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")
2011 Apr 12
0
No subject
a phone system, and plug it into a SIP Adapter like the PAP2T.
Never done it myself, so I can't recommend a suitable intercom. Hopefully s=
omeone else can.
Dan Journo
Kesher Communications (UK)
Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h=
ttp://www.keshercommunications.com/hostedpbx.html>
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
is executable.
Then run 'agi debug' from the asterisk cli, place a call and see what was
send and receive from your agi
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
I have the below
2006 Nov 02
0
testing
<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:dt="uuid:C2F41010-65B3-11d1-A29F-00AA00C14882" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<META
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.