Hi,
Ok, thanks.
/Henrik
Från: Yves A. yves...@gmx.demailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Datum: torsdag 14 mars 2013 10:48
Till: Asterisk Users Mailing List -
Top replying ...
In the CDR you have two fields, duration and billed. Duration is
the total time from Dial command to end of calls. It is the time the
Dial command is running. Billed is the time from when the other
party answered and the end of the call.
In your example, duration and billsec
Answered means your call answered by answer application or by ivr or moh
kind of dialplan.
Here, call connected to asterisk means your calls starts ringing and its
start duration field counter.
As soon as you answer the call, i,e, start playing file or moh its start
counter of billsec field.
In
Taking a look at the DEBUG statements that are associated with the
thread processing the SIP response:
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: Splitting 'FQDNz:5060' into...
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: ...host 'FQDNz' and port '5060'.
[Mar 15 13:16:08] DEBUG[27947] netsock2.c:
hi,
00:00 -- Call Connected to asterisk - duration start here
00:01 -- welcome greeting starts billisec start here
00:11 -- welcome greeting ends (10 sec wav file)
00:12 -- Call enters queue and at the same time rings on first available
extension
00:15 -- Call is answered by an agent
Thank you every one.
Now I understand why I was confused.
I have always been using Asterisk in an Inbound environment.
Hence my thought were misaligned wrt answered billed.
Now I understand. Thank you all!!
Is there anyway to capture the time for conversation, IVR, hold etc etc.
If not inbuilt
You can add custom fields in the CDR, so your dialplan can store start
time, end time and duration whenever you like.
Just use something like the
Set(CDR(customfield)=100);
Leandro
2013/3/18 RSCL Mumbai rscl.mum...@gmail.com:
Thank you every one.
Now I understand why I was confused.
I have
hi,
try Asterisk manager or AGI.
On Mon, Mar 18, 2013 at 12:36 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
Thank you every one.
Now I understand why I was confused.
I have always been using Asterisk in an Inbound environment.
Hence my thought were misaligned wrt answered billed.
Now I
Thanks,
You are right, the bash version should be:
#!bin/bash
#Get and spawn AGI variables
declare -a array
while read -e ARG [ $ARG ]; do
array=(` echo $ARG | sed -e 's/://'`)
export ${array[0]}={array[1]}
done
echo EXEC \Dial\ \DAHDI/g2/$agi_dnid\
#Get execution answer
answer=read
If you use application Queue to pass the calls to the agents you will
have the advantage of having the queue log available which will give you
lots of detailed information.
Regards
Ish
On Mon, 2013-03-18 at 17:06 +0530, RSCL Mumbai wrote:
Thank you every one.
Now I understand why I was
Hi
I want to can we use asterisk as TTS server. Which can support mrcpv2 and
ssml.
Im looking for tts server with above requirement will asterisk 1.8 is
useful for me. Any configuration available.
Any opensource tts available.
Amit--
--
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
Occasionally an agent will get a call (or more often a series of calls
in a row) where neither party can hear the other, or can only hear each
other sporadically. A MixMonitor
Is the callcenter sitting behind nat?
Sent from my iPhone
On 18 mrt. 2013, at 19:31, Mitch Claborn mitch...@claborn.net wrote:
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
Occasionally an agent will get a call (or more
Agents and Asterisk server are in the same network, behind the same
firewall, so there is no NAT between agents and the server. The outside
calls come in on a T1 fed into the asterisk computer.
Mitch
On 03/18/2013 01:44 PM, Gertjan Baarda wrote:
Is the callcenter sitting behind nat?
Sent
Hi folks,
Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
to 1.8.13, my server is no longer able to register a connection to a SIP
account at my ISP (XS4ALL in the Netherlands). At the same time, it is
still able to register a different account with another SIP
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn mitch...@claborn.netwrote:
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
Occasionally an agent will get a call (or more often a series of calls in
a row) where neither party
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