Displaying 20 results from an estimated 200 matches similar to: "Why isasterisk's voice mail calledcomedian."
2007 Sep 14
6
Force a new user to configure Comedian mail?
In Asterisk 1.4, is there any way to "force" new users to configure
their mailbox? I'm thinking a simple IVR that holds a user's hand
through changing their PIN, recording their name, and setting up one or
both greetings, the very first time they use the account.
I know I can publish docs that tell them how to use the "0" menu and do
this by hand... but users are
2005 Jun 30
7
Voicemail => SMS
Hi
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes
However I can't seem to find a way to test is a message is left. I
have tried
2007 Mar 05
1
Voicemail question
Group
In voicemail.conf I would like to having the following setup per
context not per-mailbox settings
serveremail
userscontext
fromstring
usedirectory
emailbody
pagerfromstring
dialout
sendvoicemail
callback
review
operator
volgain
nextaftercmd
forcename
forcegreetings
tempgreetwarn
Can this be done?
Thanks!
-------------- next part --------------
An HTML
2013 Jun 03
2
Difference MySQL between 1.6.x and 11.4.x
Hi
i have installed a new Asterisk server on Fedora. My first server use
Asterisk 1.6.x with a MySQL CDR and
realtime.
I have a small problems, when i configure on the new server, the same
information in MySQL, we have a error:
[Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to
connect database server SSI on myhost.myserver.com (err 2003). Check debug
for more info.
2005 Mar 21
3
US pstn => voip
Hi
I believe this is due to the way US phone systems work, however I'm going to
ask anyway. In the UK there are several providers who provide national rate
PSTN => Voip gateways which are free to receive calls on, (for the
recipient), the caller pays the cost of calling. E.g 0844 0870 etc.
I am looking for a US provier who offers the same sort of system. I don't
call the US but I
2005 Mar 29
3
Upgrade *@home to CVS-HEAD
Hi
I installed *@home 0.6 to play with the system, and learn. I tried AMP but
didn't like it and so set forth into the conf files manually. I have it all
set up how I want, all my extensions work etc. Reading this list and
playing in the wiki and google, I get the impression *@home is great for
learning, but has a few limitiations. I would like to upgrade my box to the
latest stable cvs
2006 Oct 30
3
Server Recommendations
We have a number of clients who will be needing a server to host
Asterisk on. Many of these clients use analog (FXO) lines that will
need to be connected to Asterisk via Sangoma cards. Can anyone
recommend an industry-standard server (like IBM, Dell, HP, etc.) that
has enough open PCI slots to handle up to six of the Sangoma cards? We
would like to be able to tell the customer to just go
2005 Jan 10
1
Agent Status on FOP
The hype and documentation for the last couple of releases of the Flash
Operator Panel claim that the Panel can be configured to either change the LED
for a phone, or the name of a phone to indicate when that phone is logged into
a queue. I've tried on two different versions (0.18 and 0.19) on two
different systems to get this feature to work, and have been completely
unsuccessful. Any hints
2006 Mar 29
7
Reporting?
Is there anyway in asterisk to figure out how much time an agent has
spent on the phone? I know I can see total time for a call (inbound
or outbound) but where/how do I view queue stats?
2005 Nov 08
2
networking issues
Hello all,
I'm quite new to the list, but experienced in linux. For the last several
months, I've been trying to get some basic things working in WINE (eg. Internet
Explorer).
I'm running Ubuntu (Hoary and Breezy), and debian (Sarge). It appears that no
matter what version (20040615, 20041019, 20050310, 20050930, 20050725, 0.90) I
try to install (or build and install), I don't
2005 Mar 15
6
Realtime config
Having problems getting realtime working, I'm trying to use odbc for all
of this. I've got Fedora 3 and have been fighting with odbc for a day
now. I think I got it working correctly, however I can't seem to get the
realtime portion working. In asterisk 'odbc show' shows it connected, I
see it on my (odbc) mysql server connected and all, it connects and just
idles. So, without
2006 Feb 23
0
maxmessages and maxgreet per mailbox
>From voicemail.conf:
; Maximum number of messages per folder. If not specified, a default value
; (100) is used. Maximum value for this option is 9999.
;maxmsg=100
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Maximum length of greetings in seconds
;maxgreet=60
I would like to configure these parameters on a per mailbox basis using
Realtime voicemail. I
2007 Jan 09
2
Attatching VM via email for more than one user
Hi List,
I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI:
uniqueid => 17
customer_id => 0
context => techmast
mailbox => 14
password => 1234
fullname => Sales and Service
email => user1@domain.com
email =>
2005 Mar 29
5
ACD queue question
I have a simple 4 person ACD queue using the AgentCallback function. No
matter what strategy I use, anytime someone calls into the queue
asterisk dials the agents in the order that they are listed in the
agents.conf file. This doesn't seem right to me, or am I wrong.
2015 Jan 20
1
Mailbox password change problem on realtime engine
Hello,
I am struggling with what seems a common unresolved problem, changing the
password from voicemailman when using a realtime engine (adaptive_odbc in
my case, connected to mysql).
I have seen messages dating back to 2007 with this problem and the last one
was bug 5168, reported as closed, but without explaining the fix
2013 Jan 22
2
Asterisk voicemail minimum length / silence settings
What I'm trying to achieve is that a voicemail message should be at
least 3 seconds long for it to be saved, but *after that* a prolonged
silence (e.g. 10 seconds) should terminate the call and recording.
My current settings (Asterisk 10.7.0 and 11.2.1) are:
; Minimum length of a voicemail message in seconds for the message to be kept
; The default is no minimum.
minsecs=3
;
2005 Mar 24
5
* -> SMS w/out PSTN
Hi all
I have been googling and wiki-ing and have found a number of potential
solutions to my questions, but I don't want to have to play about for too
long and risk messing up my * box now I've just got it working, if one of
you kind folk could offer your 2 penneth, (being a Brit I'll have none of
this cents business ;] ).
I want to send an SMS message whenever I get a voicemail
2015 Jun 16
4
howto copy a voicemail message to another machine ?
My asterisk server is in the cloud. Figuring out how to send an email is
too much brain damage. So i can't use the email feature that's built
into voicemail.
What I want to do is execute a remote command with the voicemail as an
argument. The remote machine command would email the message.
I'm thinking of:
same =>n,VoiceMail(vm,u)
same =>n,System(ssh myserver "emailVM
2004 Sep 09
2
Conference Phone
Any advice on a good conference phone that works with Asterisk? I like
the Cisco line and was wondering if anyone has used the 7935 or 7936
phones. From what I can tell they don't have a sip load. Has anyone
verified this or gotten an ETA from Cisco?
Chad
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2006 Oct 11
10
GPL Softphones
Hi,
I'm searching for GPLed softphones. I found WengoPhone but actually not
available for Asterisk PBX, only for Wengo network. I found Kiax but only
for IAX protocol.
Did you know a good GPLed softphones which works on Windows ?
Thanks
Greg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: