Displaying 20 results from an estimated 100 matches similar to: "After Dial execution, using DIALEDTIME, ANSWEREDTIME"
2009 Oct 30
2
DAHDI/ZAP overlap dialing
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an Alcatel prefix of type "ARS Prof.Trg Grp Seiz.with overlap".
I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which always shows
2006 May 30
1
Asterisk::AGI and DIALEDTIME
Hi List,
In one of my AGIs (using DeadAGI) I grab the answered time using:
my $res = $agi->exec ("DIAL $dialstring");
my $answeredtime = $agi->get_variable ("ANSWEREDTIME");
However this information differs from what's written in the Master.csv
file (which happens to be the correct value!)
Any ideas why?
I'm using asterisk 1.2.7.1 and the
2009 Dec 15
2
member (In use)
Hello list.
We just upgraded to 1.6.1.11.
We are using real time information stored on mysql databases. That is all
running fine.
Now, since we upgraded, some member don't get calls from queues.
In CLI: "queue show" shows something like:
611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no
calls yet
We use the extension 611 in different computers, in the
2009 Feb 21
2
DIAL() application 'g' option
Hi All,
Asterisk 1.4.12 on CentOS 5
I'm trying to increment an AstDB key with the length of the last
outgoing call. Here's what I've got for "01" UK geographical numbers:
exten => _01.,1,Dial(${UKGeographical}/${EXTEN},,g)
exten => _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME})
exten => _01.,n,Set(CALLTIME=${DIALEDTIME})
exten =>
2017 Dec 26
4
Answered time on channel
Hi,
I have a dial plan where I need to notify an external system when a call
was answered and when the call hung up. In both requests the start time
needs to be the same. My Dialplan looks something like this:
[outbound]
Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier))
Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME:
${DIALEDTIME}
2006 Jun 20
1
AGI: Dial and Recording my own CDR
Hi folks --
I have a FastAGI Perl script running, handling calls. It works great.
At one point I have a Dial() command. If the called party hangs up, Dial()
returns 0, and when I call my own recordCdr() function using the channel
variables ANSWEREDTIME, DIALEDTIME and DIALSTATUS, everything is fine.
However, if the called party picks up, and then the dialing party hangs up
Dial() returns -1,
2009 Jul 03
2
Trigger an action when B number answers the call
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2011 Sep 07
4
(no subject)
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls.
Please suggest me the solution.
[TB]
exten => _X.,1,Wait(${INCOMING_WAIT})
exten =>_X.,2,Verbose(TB)
exten =>_X.,3,Answer()
exten => _X.,4,Set(mainLoop=0)
exten =>
2009 Feb 18
1
Accumulated call time
Hi All,
Asterisk 1.4.12 CentOS 5
My ISP account includes nearly 500 minutes of VOIP calls per month but
the service is expensive for unbundled minutes. So I'm trying to find
a way to keep an accumulated total of calls made through that trunk so
that I can automatically switch to a lower-cost provider when my
bundled minutes are used. The plan is to store the accumulated time in
AstDB and
2017 Dec 27
3
Answered time on channel
It seems that what ever I set in my answer handler does not show up in the
hangup handler. In order to do billing I can't rely on the g option where
the caller hangs up the call. Looks like I can either use h or a hangup
handler along with the shared function.
On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote:
> Don't use an 'h' extension, use
2010 Jun 08
3
Limit total length of calls to a specifig SIP peer
Hi,
I'm currently using a cheap SIP provider for outbound calls.
I do have 6 channels to them.
In their terms of service there is the following limit:
The total duration of calls during one single day should not exceed 24
hours or we do have the right to terminate the contract...blah blah
What is the best way to use this provider as long as we are below let's
say 22h in a single day
2010 Aug 17
1
MySQL Connect problem...
Right, I'm baffled.
I have:
exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten => s,n,MYSQL(Query RESULT1 ${DB1} SELECT\
2000 Apr 12
1
Solaris2.6/NIS+/Samba/NT-users
Hello,
I have 20 Solaris 2.6-servers with NIS+ in different cities and countries
and a lot of NT4 servers in different domains and I use samba with nis+
support
on the Solaris-enviroment to let the NT-users access their
unix-home-directories.
The NT-clients has to send their passwords in "clear-text" to access their
directories.
Their NT-username is the same as their Unix-username
(the
2007 Oct 30
1
Size of Exten when using IAX
Hi,
We are use IAX protocol between two asterisk servers.
Now we send information through this protocol by using EXTEN
We see that the variable EXTEN only holds 66 characters.
If we set a value larger then 66 characters, for example 70 characters.
The last 4 characters are cut off.
Is there a way to increase this variable?
Kind regards
-------------- next part
2013 Aug 14
1
groupcount fraud problem
hi,
i have strange problem with call-limit/groupcount limiting. i set up
limit of 2 calls.
i'm using both methods but a for few times i have problem with
successfull fraud with more calls than 2
asterisk is 1.8.22
someone with the same problem?
any ideas how to solve or debug this problem?
--
---------------------------------------
Marek
=======================================
2015 Feb 22
2
dialplan contexts syntax and terminology
I'm looking into the dialplan specifics:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk
2004 Nov 20
1
Asterisk dead but pid file exists - gdb asterisk core.13089
Dear ALL,
Any clues or tips for the following gdb messages.
[root@localhost asterisk]# uname -a
Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct
29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux
localhost*CLI> show version
Asterisk CVS-HEAD-09/22/04-11:19:09 built by
root@localhost on a i686 running Linux
[root@localhost asterisk]# gdb asterisk core.13089
GNU gdb Red Hat Linux
2004 Aug 25
2
spandsp and certain (e.g. Canon) fax machines
Hi,
Several people have reported problems sending faxes from spandsp-0.0.1k
to Canon FAX machines. A spandsp user had the same problem with another
make of FAX machine, and traced the problem to a bug in the file t30.c
of spandsp. Line 542 says s->t4.rx_file[0] where it should say
s->t4.tx_file[0]. This fixes his problem, and I suspect it will also fix
the Canon fax machine problem.
2012 Aug 01
2
Problem with callfile and CDR
Good afternoon list.
I am experiencing a problem with the CDR and callfiles. What is happening
is this: When generating a call with a callfile, everything works
perfectly, but the CDR is recorded in the table when they answer the call
destination. The field disposition is being recorded correctly, but the
duration field is marked with the ring time and billsec is marked with 0.
This just happens
2006 Nov 22
1
DTMF detection during Call
Hi
I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by
outbound SIP.
Now i want to detect DTMF-Tone Code coming from the called party to
trigger a signal.
Can this be done with asterisk? I read that the codec with DTMF
detection are ulaw and alaw. But i couldn't find a command to detect
dtmf's within a normal call.
thanks and mani greetings
Christian