Displaying 20 results from an estimated 1000 matches similar to: "Problem Transferring Direct to Voicemail"
2007 Feb 16
5
FW: Problem Transferring Direct to Voicemail
Skipped content of type multipart/alternative-------------- next part --------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
2006 Dec 12
4
MeetMe Conferencing and Marked Mode
I am trying to set up a Conference room where users are put on hold
until the host arrives. I have figured out that the A option activates
marked mode and the w option is used to activate the waiting until the
marked user arrives. This seems to be what I need. What I can't seem to
find is how do I mark a user?
Thanks
_____________________
Kevin Savoy
Business Unit Telecom Analyst
2218 4th
2007 May 01
3
Display Caller ID of called party
Not sure if this can be done or not, but I can't seem to find it
anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I
would like to have the caller id of the person I am dialing displayed
and not the number I just dialed. Is this possible? So, if extension
4023 is John Doe, and I dial 4023, my display should read John Doe and
not 4023. I am using a Polycom 501 by the way in
2007 Feb 07
1
After upgrade to 1.4 transfers don't work properly
I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I
have confirmed this on multiple phones. When the called person answers
and tries to transfer the call to another extension, the call
successfully transfers, however the person answering the transfer cannot
hear the person that called in, the caller. My
2007 Apr 13
1
Call Recording Servers
We are looking at using Asterisk as a call recording server for an Avaya
VoIP S8700 system in a multi-site VoIP Call Center. All calls will be
coming in to one location and sent out via VoIP to other call centers.
What kind of specs should we be looking at purchasing for our Asterisk
server to be record up 200-300 calls simultaneously?
Linux runs in 64 bit architecture, but does Asterisk
2007 Feb 12
1
FW: After upgrade to 1.4 transfers don't workproperly
Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it actually posted.
The below worked for normal transfers. Now here is another situation. When we try to transfer a call directly to voicemail it plays the voicemail message but we can't transfer the call. The only way I could get it to work was to do a conference and then drop out of that conference.
My
2006 May 05
6
Dumping queue_log to MySQL
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 2151 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060505/1729faf8/attachment.gif
2007 May 26
4
reset Polycom phones remotely
I have provisioned a bunch of Polycom 301 phones to get the config files
from my ftp server. Out of the 4 phones 2 get the config file however the
other 2 cannot contact the boot server. I have reboot the phones a number
of times remotely (the client is 400 km away) through vnc and logging onto
the web config internally. No matter what I change on the web config page
it is not saved. I feel I
2006 Dec 28
1
FW: cdr_addon_mysql.so did not register itself duringload
So no one else is having issues with MySQL and 1.4? I'm the only one?
-----Original Message-----
From: Savoy, Kevin - Williston, ND
Sent: Wednesday, December 27, 2006 2:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] cdr_addon_mysql.so did not register itself
duringload
Well the addons from 1.4 are installed. This original Asterisk
2007 Feb 08
1
After upgrade to 1.4 transfers don't workproperly
This worked. Great and thanks
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
On Wed, 2007-02-07 at 14:12
2006 Dec 28
2
FW: cdr_addon_mysql.so did not register itselfduringload
Ok so something is missing. I get the below for those two lines.
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... no
I even installed the mysql-devel as Bradley Watkins suggested and still
it says no. What do I need to make that say yes?
Thanks
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
2007 Jun 18
2
MixMonitor Timestamp problem
hi,
I am facing some issues while using MixMonitor. My
extensions logic is attached below:
exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b)
in this extensions TIMESTAMP is not working in Asterisk 1.4. can any
help me why TIMESTAMP is not working in Asterisk 1.4.
regards,
Asif
2009 Feb 13
3
MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
Hello Asterisk Users and those with an Interest in VoIP Tech,
The agenda is open for our next meeting. I think we'll plan on an open
discussion of anyone's choosing. If we're lacking a topic, we'll give a
demo of installing fail2ban for your asterisk system.
Bring your questions, ideas and projects and we will help you work through
them.
Jimmy John's is just a block away
2006 May 05
10
Call Center Phone with Auto Answer
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 2151 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060505/ced665b2/attachment.gif
2006 Dec 26
1
cdr_addon_mysql.so did not register itself during load
I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4
as well. I can place calls but I noticed the MySQL was writing out to
the database. When doing an Asterisk load with asterisk -vvvv I saw the
following:
[Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module:
Module 'cdr_addon_mysql.so' did not register its
[Dec 26 11:02:08] WARNING[10029]:
2007 Jan 17
3
Asterisk 1.4 and CDR
Hi guys, I have recently installed a Asterisk Server with CDR Call Detail
Records. I have installed it over a Asterisk 1.2 , but now It do not run
. I have installed it with the following procedure:
# yum install ncurses
#yum install openh323-devel
# yum install mysql-server
# yum install mysql
# yum install php-gd
# yum install php-mysql
# yum install mysqlclient10
# yum install zlib
# yum
2011 May 04
4
Syntax Help on a Bash Script
Hi All,
I'm brand new at doing anything linux and would like feedback on this
script I'm trying to understand from an example I'm working on..
Oh, running Centos 5.6
Anyhow, I run this bash script:
#!/bin/bash
# send data to the table in the MySQL database
MYSQL='which mysql'
if [ $# -ne 4 ]
then
echo "Usage: mtest4 empid lastname firstname salary"
else
2006 Apr 26
2
Status of Queue
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 2151 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060426/6367e36a/attachment.gif
2011 Oct 27
5
Asterisk Executing outbound dial number twice
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
== Extension Changed 4773[sipphones] new state InUse for Notify User 4701
-- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu
option. If they just sit at the main menu, after 20 seconds, they are
transferred to the operator. If the user picks an extension from the
directory, they are transferred to the proper extension. If the called
number is not available, they are transferred into VoiceMailMain. They
leave a message, and hang up. The hang