Displaying 20 results from an estimated 6000 matches similar to: "Variable Scope Question"
2006 Apr 04
3
Auto Attendant Question
Hi Folks
I have had a look through the Features list, and I see that the system does
support an auto attendant, however is it possible to have say 5 telephone
numbers that a person would dial and have 5 different messages
I.e
Dial 555-1121 and you get a message for companyA call centre
Dial 555-1131 and you get a message for companyB call centre
Dial 555-1141 and you get a message for companyC
2012 Oct 18
1
Samba4 - multiple forest hosting?
Can I configure Samba4 in such a way that I have two separate **forests**
on a single machine? let?s say one for CompanyA and other for companyB?
So essentially does Samba4 support multiple server instances like Samba3 as
described here http://wiki.samba.org/index.php/Multiple_Server_Instances?
If it does not yet, are there any known blockers in supporting this or it?s
just a question of time
2004 Nov 08
5
Same Extensions in Multiple contexts
Hi
For a customer, I am trying to setup 3 different companies on one asterisk
box, and I need to assign extension 200 in three different companies. I was
using different contexts, but was unable to get it to work. So, my basic
question is -
In Asterisk, Can we have same extension number in different contexts?
For example:
[Context_company_1]
exten => 200,1,,,
2009 Aug 01
3
Dialplan strategy suggestions needed
I have a new Asterisk system going into production next week and I'm a
bit stumped as to the best way to handle the Dialplans for it.
The Asterisk system is replacing 4 separate PSTN lines with both SIP &
PSTN inputs. The setting up of the dial plan is giving me some design
headaches, which probably means I'm missing something obvious and doing
this the hard way.
I have separate
2005 Sep 28
1
Tiny Echo on PSTN via Zaptel
I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino
channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog
phone on the inside connected to one of the FXS ports, and PSTN line
connected to one of the FXO ports.
My problem is that as soon as I hear the _first_ ring when I dial out
through the PSTN line, I hear a tiny echo on the phone (I estimate
between
2005 Oct 04
5
PBX 'Personalities' ?
We are running our * server as a virtual PBX for 6 companies. I am having
all of the Allison prompts plus our own custom IVR prompts being re-recorded
for each company, in a different voice (marketing thing) with a different
personality (perky, corporate, earthy) .
I'm curious if someone could point out a dirty trick to get the voice to
play right, for internal and external callers,
2003 Sep 04
2
Incoming CallerID management
Greetings,
I need if possibile an explanation on how to manage the incoming callerid
for an incoming call. Let me explain the situation:
We have two different companies in this office that shares the same PBX (*
box). Each company have its own number for the incoming calls.
What i'd like to implement is something that, depending on the incoming line
that is involved in the call, plays a
2004 Sep 02
5
Any way to _always_ execute certain commands in a dialplan context?
I've got a need to do something like the following:
[foo-context]
exten => _.,1,SetCIDNum(123)
exten => _.,2,SetCIDName(XYZ)
include => local
include => tollfree
But of course, this example won't work. The goal here is this: if a call
ends up being handled by the "local" or "tollfree" contexts, I want
those SetCID*** commands executed. Otherwise, I
2004 Jun 11
3
Background Playback fails
Hi Guys.
I've had a lay off from Asterisk for 12 months but I am starting to look
into it again. I am not very Linux savvy and found it hard going the
last time. I've started playing with it in the last 3 weeks and I have
to admit to making more head way this time.
The first problem I'm stuck on and I cant find a solution to is that
sound files that I have recorded (be it by
2008 Oct 02
1
Asterisk Queue question
When the asterisk a queue reset their counters?
I 'm talking about this kind of info in asterisk console.
>show queue 600
600 has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s
I just say that because I have a queue with strategy "Fewest Calls"
working for a couple of mouths, and a new agent has been added this
2006 Apr 05
5
Dial Plan Logic Problem
Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten =>
2008 Nov 21
2
Log level of 500 Server Internal Error.
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.
I wonder if such SIP fails could generate at least WARNING in log?
Currently i'm checking logs for warnings and errors, so i probably
have missed those.. It would be
2006 Nov 19
2
WaitExten only reading 1 digit.
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
"Waitexten" app is failing to read 3 digits and just reading the first
and then announcing that it is invalid since all extensions are 3 digits.
How do I make Waitexten wait for 3 digits?
I have setup the extension "100" for users to reach the
2008 Oct 06
1
AEL and swap from macros to contexts
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c
2007 Oct 17
3
Play sound on hangup
Hi,
Does anybody have some ideas - how to play a sound file on channel, after that
bridged channel got hanged up?
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
2007 Sep 12
2
Callback for unanswered transfers...
Hi,
Does anybody know if there is a way for a call goes back to transferer if
unanswered ?
Thanks
Luis A P Barbosa
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2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2005 Jul 23
2
(cause 66 - Channel not implemented) -- IAX?
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and updated just now.
The error is:
-- Executing Answer("Zap/1-1", "") in new
2007 Sep 11
3
Prevent multiple sip registrations
Hi all,
Is there anyway i can prevent multiple sip registrations from different IPs
using single username in asterisk. Does asterisk provide any aid in this
respect? As far as my knowledge is concerned i dont think there is any
support for this in asterisk, so i think i'll have to makeup a script which
sniffs sip packets coming for asterisk and detect for multiple register
requests coming from
2008 Feb 18
2
SiP call generator
I want to have a PC-based real-time VoIP bulk call generator (including both
SIP signaling and RTP generation)
for stress testing and precise analysis of the VoIP network equipment.
Do any one knows a free program can do that .
Regards
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