Displaying 20 results from an estimated 800 matches similar to: "No subject"
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
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
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
2003 Dec 01
0
No subject
entries)
[2002/06/07 12:46:16, 0] smbd/service.c:(614)
1001627gefage (3.61.235.23) Can't change directory to /tmproot (Permission
denied)
[2002/06/07 12:46:17, 0] rpc_client/cli_netlogon.c:(157)
cli_net_auth2: Error NT_STATUS_NO_TRUST_SAM_ACCOUNT
[2002/06/07 12:46:17, 0] rpc_client/cli_login.c:(74)
cli_nt_setup_creds: auth2 challenge failed
[2002/06/07 12:46:17, 0]
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
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>
2011 Sep 02
0
No subject
1. Does "Wrap-Up-Time" apply to all queue agents/extensions that just rang,=
or only the one who actually answered the call (I assume the latter)?
2. Does the "Member Delay" delay the ringing of new calls to agents, or onl=
y come into play AFTER the agent answers the ringing call?
Any other suggestions for how I can resolve this issue? I am wondering whet=
her "Agent
2010 Jul 03
0
No subject
eeds to be added to the database or coding of the program. Is this right? I=
sthere something else I need to do??
Ray Garza
System Analyst II
Materials Management
(956) 381-2163
--_000_60EAA3B47B099544A16557729563A1D608920F4FCCINFDBMBX4dsut_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
<html
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
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
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"
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.
2011 Jan 10
0
No subject
------=_NextPart_000_00BF_01CBCF4C.8B374370
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" =
xmlns=3D"http://www.w3.org/TR/REC-html40">
2011 Jan 10
0
No subject
major undertaking. But since you are using an AGI to control the Queue
command instead of using it from the dialplan, you have more control over
this problem than you realize. For simplicity of illustration, let's say
your AGI simply wants to take a call and send it to the next agent in the
queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the
Polycom transfer from
2012 Dec 26
2
dovecot crashing?
Happy holidays! I am experiencing an issue when trying to check my mail
using IMAP. with Dovecot I have tried checking my mail using a full GUI
client (Thunderbird) and telnet. Both times I get disconnected before all
of my messages can be downloaded and I see an error in my mail log. Here
are the details:
[root at cust19-1-prod-domain userqa]# dovecot --version
2.0.9
[root at
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 Jul 20
0
No subject
I am looking for status of each number dialed out.
Whether its failed or successful .
Any way ?
------=_NextPart_000_00A1_01CA27F6.DE850CA0
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: [asterisk-users]
2009 Jul 20
0
No subject
I am looking for status of each number dialed out.
Whether its failed or successful .
Any way ?
------=_NextPart_000_0124_01CA2A21.42AA12D0
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" =