Anyone has an idea of implementation ?!
Jonas.
On 08/10/2010 09:04 AM, Jonas Kellens wrote:
Hello list,
situation :
1. incoming calls come into a queue
2. there is 1 agent logged in into the queue (not always the same agent)
3. when the caller is in the queue, he has the option to quit the
Hello.
I notice that when a call that is recorded with MixMonitor is transfered
to another co-worker, the recording ends.
exten = 409,n,Macro(SDstartrecording,external,${DID})
the incoming call then goes to a queue...
[macro-startrecording]
; ARG1 = incoming DID or CALLERID(name)
; ARG2 =
I tried it but I still cannot hear any sound created from Festival()
function. I can hear only a voice saying one which was working earlier as
well. Here is log of asterisk console:
-- Attempting call on SIP/011xx...@gafachi1a for
s...@connect-to-me:1 (Retry 1)
-- Executing
Hello list,
I've been having a problem for some time now that I can't figure out how
to solve it.
On a PTP BRI ISDN line, if I have both channels in use and I place a
third call from the outside, I'm not getting a busy tone like I should.
Instead I get a congestion tone, as if the line was not
Nick Brown wrote:
Depends what its connected to
True,
But for most people, that would be the correct answer.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
I just noticed that if I change the dialplan (connect-to-me) to following
(shifting places of SayDigits and Festival() ):
[connect-to-me]
exten = s,1,Answer
exten = s,n,Festival(hello john)
exten = s,n,SayDigits('1')
exten = s,n,Hangup
I do not hear any voice, not even anyone saying ³one². It
On Wed, Aug 11, 2010 at 02:04:20PM -0400, Jerry Geis wrote:
I have DAHDI 2.2.1
my system.conf file is :
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
echocanceller=mg2,1-23
loadzone= us
defaultzone = us
Its telling me:
Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1:
So in broad terms
You need to know when the queue is empty, and when there is voicemail (in a
generic queue mailbox presumably) and also that you haven't already delivered
the voicemail, and probably that when you deliver the mail its been
successfully been heard and actioned.
Are you also
Does anyone knows a way to connect skype to asterisk?
I know digium and skype have a way to do it, but it's not free?
I read something about siskyee, anyone has ever tested it or know other free
app to connect them?
thanks!
--
_
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
top/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
Next step: start tracing.
sh -x /etc/init.d/asterisk start
--
h -x
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Hi list,
I was going through my dialplan today and found these 2 oddities
with meetme using DAHDI to join the conference.
1. Although music on hold is indicated, I don't get any sound until I press
*. Then the conference menu plays and all is well except -
2. According to the
Without diving into too many details, does anyone have a simple callback
script that does the following:
Caller -- Dial
Asterisk -- In order to place this call please enter a callback number
to place this call for your pin...
Caller -- Enters DID to call back for pin
Asterisk -- stores a number
Danny Nicholas da...@debsinc.com wrote:
Hi list,
I was going through my dialplan today and found these 2 oddities
with meetme using DAHDI to join the conference.
1. Although music on hold is indicated, I don't get any sound until I press
*. Then the conference menu plays and
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Subject: Re: [asterisk-users] Problems with meetme in 1.4.26
Danny Nicholas da...@debsinc.com wrote:
Hi list,
I was going through my dialplan today and
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Subject: [asterisk-users] Callback script anyone
Without diving into too many details, does anyone have a simple callback
script that does the following:
Caller -- Dial
Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Subject: [asterisk-users] Callback script anyone
Without diving into too many details, does anyone have a simple callback
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Sent: Thursday, August 12, 2010 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback script anyone
Danny Nicholas wrote:
A client asked me to come with a system that will pass certification with
PPP in the UK.
Google is not being helpful :(
It has something to do with recording calls in case the PPP requests a
copy. Supposedly their rules were relaxed on August 1st if that helps.
Any clues (links?) will be
hi,list
i installed App_Konference in my Asterisk 1.6.2.11.
and i write in dialplan like this:
exten = 95040,n,konference(1234,RVxTH)
it works fine. but I want to record the conference, if use MeetMe , i
can use 'r' option to do this.
but there is no 'r' option in konference , Could you
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