Displaying 20 results from an estimated 3000 matches similar to: "Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered"
2009 May 18
5
LDA and postfix with recipient delimiter: documentation
Just a comment about the documentation for postfix integration when
using receipient delimiter and delivery of user+foo at domain.ext to
user at domain:
http://wiki.dovecot.org/LDA/Postfix
I had some issues to get this working, so thought I'd share my
experience. Dovecot version 1.2.rc3. Postfix version
postfix-2.7-20090511.
I have no local (unix) mailboxes, everything is delivered to
2007 Oct 24
2
How to tune Asterisk AMD - Answering Machine Detection "hacks"
Hello Everyone,
Can someone point me to reliable links on how to tweak Asterisk AMD
I am calling a number and have to two files to play depending if it is a
real person or an
answering machine.
Most everytime Asterisk calls it thinks it is an Answering Machine and it
starts playing
the AMD message, instead of the delivering the "1st real message"
Any hints?
Thanks in advance,
-C
2006 Nov 12
0
Trixbox dialout problems
Hello All.
I am trying to use RAGI the ruby agi framework with trixbox. I am
having a problem
with the dialout part. The RAGI framework creates a file in the
/var/spool/asterisk/outgoing directory and routes the call to an
extension (I have listed the relevent portion of the file below). The
problem is that the initial dial command does not execute properly in
trixbox. I am hoping somebody who
2006 Jan 21
3
need some help designing my threaded messaging system
Hi,
I want to create a messaging system that recognizes threads of messages, not
unlike gmail.
So far I have these models:
Conversation
belongs_to :user
has_many :messages
Message
belongs_to :conversation
The problem I am running into is not only does a conversation belong to a
user but the conversation also has a receipient user with his/her
corresponding conversation. How would I
2005 Jul 23
2
ASTCC gives me only the time, but no cost
I try to track down an error that causes that Astcc just reports the time, but not the costs.
I could narrow the problem down into this sub routine:
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment };
my $cost;
print STDERR "Adjusted time is $adjtime, cost is $adjcost with
2008 Aug 21
3
After Dial execution, using DIALEDTIME, ANSWEREDTIME
Hi,
I noticed that when dial terminates it does not return to the dialplan,
and therefore can not execute any entry after Dial().
Is there any trick to overcome this limitation ?
How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if
I can not execute anything after Dial()?
I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls
end
2010 May 03
1
BADTIME FOR ANSWEREDTIME
Hello,
I saw that Asterisk don't calcultate fine the ANSWEREDTIME.
I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10
because, now, if ANSEREDTIME =~ 15.9, it become 15! it isn't correct
How can I have a rounded ANSWEREDTIME ?
Where have I to manipulate the sources?
thank you
--
Francois
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2007 Feb 20
2
Help! How to get ANSWEREDTIME after DIAL a ZAP channel?
Dear all,
I tried to make a call with extensions.conf.
exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten=> _00[1-9].,102,Hangup
But the 2 and 102 will not be executed.
So I can get the correct answered time via 2.
Is any idea about it?
Is it the problem of my ZAP channel's configuration?
My zapata.conf is as below:
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
2007 May 05
1
${ANSWEREDTIME} Broken on 1.2.13?
No matter what I do, ${ANSWEREDTIME} is always 0, even on the most
simplest dial plan such as:
Using Asterisk 1.2.13
exten => 77,1,Answer
exten => 77,2,Playback(custom/dax/S300) ; one minute file
exten => 77,3,Noop(${ANSWEREDTIME})
exten => 77,4,Hangup
-- Executing Answer("SIP/5402-b7b45f58", "") in new stack
-- Executing
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}
2004 May 25
1
dialout=fromvm
If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out?
Is there a service feature code?
2003 Jul 02
1
Dialout Lines ???
I've been reading the Linejack strikes again messages, and have another Newbie question
is it possible to use a Voip Product as a Dialout line for * ?
I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box?
The Vega100 does either sip or h.323.
Thanks.
Bradley Greep
2003 Oct 06
2
ISDN Dialout
Hi,
I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card.
When in Minicom, the only way I can dialout is if i issue ATS18=1 First.
Otherwise I get a BUSY message. So thats fine.
But when I dialout from asterisk, I get an immediate hangup, so my guess is
that asterisk is not issuing ATS18=1 to the ttyI device.
Here are my configs, any input would be greatly appriciated.
2005 Aug 28
0
way to prevent voicemail dialout/callback from 'outside'
I am trying to find a way to allow dialout from voicemail when connected
from an 'internal' extension context, but prevent dialout when connected
from an 'external' extension context.
As far as I can tell the dialout context that can be set in voicemail
has no regard for the context from which the call to voicemail came in.
Any ideas on this? Maybe a variable passed when
2007 Mar 21
5
automated dialout detect forward
Hi!
I have an automated dialout via a call file to a mobile.
Can I detect when the call is not answered but forwarded to the mobile
operator voicebox?
I would like to stop the dialout if this is the case.
TIA,
Mike
2004 Apr 01
1
dialout with chan_capi
Hi,
When I try to dialout over chan_capi everything works fine
when I settle for
msn=* in my capi.conf and use the primary msn of my ISDN-line.
But trying to configure a different MSN the chan_capi doesn't dial
and comes with:
No one is available to answer at this time
What can be the prob?
--
Thanks,
Marc aka IzNoGood
2004 May 18
1
Linejack dialout
Dear all
I read on the list back in 2003 that * does not support IXJ LineJACK
dialout yet
is this still the case?
Thanks
2005 Feb 27
1
dialout with PPP on ISDN to an ISP
Hello my name is Ilija Poznic and I have a problem.
My configuration is
1. Digium TDM4000P with one FXS.
2. AVM Fritz ISDN adapter (configured with capi).
When I connect to my ISP and then start *. Asterisks is registering me to SIP
provider iconnect. After that I can call international call trough VoIP.
My problem is that I want to dialout to ISP only when I have a international
call.
2010 Jul 08
1
AGI get full variable
Dear All,
I have "get full variable" AGI call to get the ANSWEREDTIME channel
variable. I have originated the call to one extension, once answered I have
called DeadAGI to control the call.
I have problem that after hangup the call AGI "GET FULL VARIABLE" returns
-1 for ANSWEREDTIME channel variable.
What is the problem? Where I made wrong. Please suggest me..