2009/12/18 Barry Miller
> On Fri, Dec 18, 2009 at 12:23:22AM +0100, Olivier wrote:
> >
> > Today, this IVR is using function AEL GotoIfTime in several places.
> > The problem is if it's 11pm at the moment I'm testing this IVR, I can't
> > nicely test the 9am or 2pm branch.
> >
> > Suggestions ?
>
On Thu, Dec 17, 2009 at 07:19:49PM -0600, Tilghman Lesher wrote:
> On Thursday 17 December 2009 17:23:22 Olivier wrote:
> > When I was testing an IVR, I realized I miss a function I would call
> > GotoIfTimeWithOffset.
> >
> > Today, this IVR is using function AEL GotoIfTime in several places.
> >
On Thu, Dec 17, 2009 at 08:29:55PM -0500, Bruce Nik wrote:
> Hello Everyone,
>
> I am making a simple index.php file which will allow a web user to enter his
> $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
> Following is the index.php and the contents of extensions_cus
Hi!
Trying to understand how wrapuptime is working...
I have written a small php script that let agents log in/out
off a queue. That part is working as a clock but wrapuptime
is not doing what I expect.
Input Interiör - Queue Manager
0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy
Steve Edwards writes:
> Wouldn't a "set time" function be more usefull?
I really like that idea. Enough that I could try to lobby internally for
funding, if you know someone who is willing to do the work...
/Benny
___
-- Bandwidth and Colocation Pr
Hi there,
I have some problems when using READ() statement in the dialplan to
collect DTMF digits.
I'm using the following within my extensions.conf to receive 6 digits
exten => 9070,n,Read(CONFNO,conf-getpin,6)
So far it works! The user getting the announcement and asterisk waits
for 6 dig
A new patch has been made for an extra Manager Event when a call-pickup has
occurred.
There are two possible situations
1) by using *8
2) by using *8123 (to pickup extension 123 when it is ringing)
The manager event looks like:
Event: Pickup
Privilege: call,all
Channel: SIP/ast163-000c
Unique
Where do you get FFA? I have not seen this, what is the minimum version of
Asterisk that you need? Sorry about the questions.
Thank you and have a nice day,
Anthony Francis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Hi,
Could someone tell me where are the good places in chan_iax to put trace
points when I experience strange delays in NEW processing?
I tried to output some debug after every stage of socket_process / case
IAX_COMMAND_NEW, but it all takes max 30ms. However, sometimes in a
normal call I get a
Obvious debugging steps at this point:-
1. Manually connecting to AMI and testing the commands
1. telnet 127.0.0.1 5038
2. Login command
3. Originate command
2. Use a dummy AMI listener instead of actual asterisk to see if the
events are going through
1. Stop asterisk
Hello!
I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We
are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino
CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950
server with 8 gigs of RAM. The 2 other ports on t
Hello,
i have a problem with a Sip trunk to a SAP-BCM PBX.
In and Outbound Calls works fine but when the SAP tries to transfer an
inbound call to an outbound call there is no-way-audio. Two outbound
calls could be transfered without any Problem.
In the sip trace i see that the SAP BCM make som
On Fri, Dec 18, 2009 at 06:53, Jason Martin wrote:
> Hello!
>
> I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks.
> We are an outgoing call center with 30 internal analog phones hooked up to 2
> Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a
>
As it is done today, the wrap-up time is not terribly useful in Asterisk, as
it is fixed-length. If you need to implement it in a real-life scenario, it
would be better to pause the agent when the call is through and have him
unpause manually when he's done the wrap-up; this way you get a measurab
Thanks for the input.
Once the form is submitted the variables obtain a value which is the phone
number, dial number, and spoof number. I did the telnet and it works fine.
Now, I am really not a php coder and just stealing code from here and there
to make this working. Here is what I noticed that
On Friday 18 December 2009 04:11:36 Tzafrir Cohen wrote:
> On Thu, Dec 17, 2009 at 07:19:49PM -0600, Tilghman Lesher wrote:
> > On Thursday 17 December 2009 17:23:22 Olivier wrote:
> > > When I was testing an IVR, I realized I miss a function I would call
> > > GotoIfTimeWithOffset.
> > >
> > > Tod
Hi all,
the aims of this mail is to use saynumber fonctionality during Music On Hold
while dialing.
Music On Hold can only play a music file
So Is there a way to pre-record the saynumber output and other .gsm file and
then play the record file during Music On Hold ?
all solutions are welcome...
If you have SOX, LAME and time, you can do about anything you want. The
default moh files are wav, but a lot of folks use mp3 with the mpg123
player.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars
Sent: Fri
On Fri, 18 Dec 2009, Tilghman Lesher wrote:
> The syntax is actually:
>
> Set(TESTTIME()=2009-12-25 10:35:00 CST)
1) Does this set the "time" to a fixed value or does it set the time at
the point of execution and then the "clock" increments from there?
2) Does this only affect gotoiftime() or d
El 18/12/09 11:31, Bruce Nik escribió:
> I am amazed that there is absolutely no proper documentation on how to
> connect to Asterisk AMI with PHP. All tutuorial just mention: pass Action:
> originate Channel: SIP/1234, blah blah blah and never give a simple example
> of php.
>
http://phpagi.
Hi Danny,
I've already have a look to those tools, but I cannot see how I can record
the saynumber output audio into a file
Mickael
2009/12/18 Danny Nicholas
> If you have SOX, LAME and time, you can do about anything you want. The
> default moh files are wav, but a lot of folks use mp3 wit
Saynumber just does an "EXECUTE BACKGROUND" on the files in
/var/lib/asterisk/sounds/digits. So to "record" a saynumber output of 23 to
a moh file, you would do sox /var/lib/asterisk/sounds/digits/20.gsm
/var/lib/asterisk/sounds/digits/3.gsm /var/lib/asterisk/moh/23.wav. If your
moh processes rand
Greetings all-
I've got a rather odd situation and would like to know if anyone can shed some
light on the issue.
Some background- I've got an * system running 1.4.11 (yes I know it's older..
upgrades are planned at some point...). I also have a remote user with a
cordless phone connected to a
On Friday 18 December 2009 11:17:24 Steve Edwards wrote:
> On Fri, 18 Dec 2009, Tilghman Lesher wrote:
> > The syntax is actually:
> >
> > Set(TESTTIME()=2009-12-25 10:35:00 CST)
>
> 1) Does this set the "time" to a fixed value or does it set the time at
> the point of execution and then the "clock
The Asterisk Development Team has announced the release of Asterisk 1.6.0.20.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.20 resolved several issues reported by the
community, and would have not been possib
The Asterisk Development Team has announced the release of Asterisk 1.4.28. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.28 resolved several issues reported by the
community, and would have not been possible w
The Asterisk Development Team has announced the release of Asterisk 1.6.1.12.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.1.12 resolved several issues reported by the
community, and would have not been possib
Dear All,
I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by
Vitelity. When the number is called it goes to my Asterisk box. The
protocol is SIP. This all works just fine if I answer the call and
begin a playback.
I want to let the number ring for a few seconds before it
The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and
Asterisk-Addons 1.6.2.0. These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.0 is the first feature release since Asterisk
1.6.1.0
Is the new Fax For Asterisk being released in conjunction with this
release?
Thanks,
--Warren Selby
On Dec 18, 2009, at 4:59 PM, Asterisk Development Team wrote:
> The Asterisk Development Team has announced the release of Asterisk
> 1.6.1.12.
> This release is available for immediate dow
Warren Selby wrote:
> Is the new Fax For Asterisk being released in conjunction with this
> release?
If it's not already available, then it will be available very early next week.
Leif Madsen.
___
-- Bandwidth and Colocation Provided by http://www.ap
How does Fax for Asterisk work?
On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen
wrote:
> Warren Selby wrote:
>> Is the new Fax For Asterisk being released in conjunction with this
>> release?
>
> If it's not already available, then it will be available very early next week.
>
> Leif Madsen.
>
> ___
http://www.digium.com/en/products/software/faxforasterisk.php
Thanks,
--Warren Selby
On Dec 18, 2009, at 7:11 PM, Thomas Perron
wrote:
> How does Fax for Asterisk work?
>
>
> On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen
> wrote:
>> Warren Selby wrote:
>>> Is the new Fax For Asterisk being
Try putting the wait before the Answer.
...
exten => s,n,Wait(10)
exten => s,n,Answer
...
On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither wrote:
> Dear All,
>
> I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by
> Vitelity. When the number is called it goes to my Asterisk box
Thanks for the input. I have explored phpagi from sourceforge but they
don't have any documentation or it's poor. I am just wondering if any can
contribute their working php file with me that does Originate action.
Thanks
On Fri, Dec 18, 2009 at 12:24 PM, Alex Villacís Lasso <
a_villa...@palosa
At 03:14 PM 12/18/2009, you wrote:
>The Asterisk Development Team has announced the release of Asterisk
>1.6.2.0, and
>Asterisk-Addons 1.6.2.0. These releases are available for immediate
>download at
>http://downloads.asterisk.org/pub/telephony/asterisk/
And any plan for Skype for Asterisk?
Ira
On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
> Try putting the wait before the Answer.
>
> ...
> exten => s,n,Wait(10)
> exten => s,n,Answer
> ...
Thanks Steve. I tried that:
> On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither wrote:
> > Dear All,
> >
> > ...
> > exten => s,n,Answer
On Fri, 2009-12-18 at 19:43 -0800, Ira wrote:
> At 03:14 PM 12/18/2009, you wrote:
> >The Asterisk Development Team has announced the release of Asterisk
> >1.6.2.0, and
> >Asterisk-Addons 1.6.2.0. These releases are available for immediate
> >download at
> >http://downloads.asterisk.org/pub/tel
At 08:31 PM 12/18/2009, you wrote:
>On Fri, 2009-12-18 at 19:43 -0800, Ira wrote:
> > At 03:14 PM 12/18/2009, you wrote:
> > >The Asterisk Development Team has announced the release of Asterisk
> > >1.6.2.0, and
> > >Asterisk-Addons 1.6.2.0. These releases are available for immediate
> > >download
If you try just this, what does the caller hear? It should be ringing
for the first 20 sec, and then maybe the congestion tone afterwards.
exten => s,1,Wait(20)
exten => s,n,Hangup
You shouldn't need/use the Ringing() command at all, as the initial
ring before your system answers would be generate
Ira schrieb:
> At 08:31 PM 12/18/2009, you wrote:
>
>
>> On Fri, 2009-12-18 at 19:43 -0800, Ira wrote:
>>
>>> At 03:14 PM 12/18/2009, you wrote:
>>>
The Asterisk Development Team has announced the release of Asterisk
1.6.2.0, and
Asterisk-Addons 1.6.2.0. These releases
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