Displaying 20 results from an estimated 50000 matches similar to: "Increase Volume on AGI"
2008 Feb 22
2
AGI / Voicemail Que
Hello All,
I have my own AGI script running and I am trying to push the call to
voice mail when Busy, Unavailable and Not Answered.
Everything is working fine but the only problem is voice mail greetings
for Busy and Unavailable is not played. By default only "Temp Greetings"
voice mail greetings is played. I am passing the correct parameters for
Busy => 'b', Unavailable
2007 Jun 27
2
Error While Calling AGI
Hello All,
Found some strange problem while Asterisk trying to call the AGI file.
If I pick up the call on the first attempt, it will execute my AGI file
properly.
But if I don't pick up the call and let Asterisk call me again, it adds
StartRetry next to my AGI file name.
Which will cause the AGI to fail to execute.
-- Attempting call on SIP/5181 for application AGI(recordvoice.php)
2005 Feb 23
4
Vonage <---> Asterisk Working Config!
Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk@home sourceforge forum, you'll probably be able to work out
how to set it up from there.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To:
2007 Aug 29
1
Monitor System using AGI Scripts
Hello All,
Anyone using AGI scripts to monitor their systems?
Something like if the system goes down, AGI script will be triggered and
system admin will be notified saying "System XYZ has gone down"...
Any suggestions...
Cheers,
Nitesh
2007 Jun 20
1
Asterisk RealTime
Hello All,
I manage to configure Asterisk RealTime and now it loads the SIP
users/peers from MySQL DB. The table I am using is of A2Billing DB
"cc_sip_buddies".
Now the only problem I am facing is incoming calls are failing... The
ATA which is assigned this DID number is behind NAT and according to
Olle's explanations he said "*there's no support for NAT keep-alives
2005 Jul 18
5
TDM04B - Takes long to initialize...
Hello All,
I got my TDM04B card installed and configured.
Everything works fine I can receive calls and route to appropriate
extensions.
The only problem I am facing is Slowness.
When I dial the PSTN number which is connected to Zap 1-1 after two
ring it answers and then run the AGI script. What I did was assign it
to a specific extension. So all inbound call on that PSTN number
should
2007 May 23
16
WiFi SIP phones
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices.
I assume things must have moved on somewhat since
2005 Mar 26
5
Click-to-Talk with Asterisk?
Hi Nitesh,
Take a look at this
http://www.microappliances.com/site/html/index.php?section=Products&page
=clienthowto.php
I've never implemented it though so I would appreciate some feedback on
if it works.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Saturday,
2005 Feb 16
2
Monitor does not like variable subsitutions
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just
wav.
Here is what I get if I just supply a filename
directly (it works fine):
--context-----------------------------
exten
2010 Mar 26
1
SIP/2.0 403 Forbidden
hi,all
when i send a call to other server use SIP trunk,
i got this below,
<--- SIP read from 222.46.18.52:5060 --->
SIP/2.0 403 Forbidden
what's rong with is?
> Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
> Created by Mark Spencer <markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
2008 Jan 18
2
SAY TIME + PHPAGI + Timezone
Hello All,
Is there any way to change the timezone on the fly? I have this little
time clock program running on Asterisk system developed using PHPAGI.
Currently, whenever user logs in, Asterisk will prompt the current
system time using "$agi->say_time();" which executes "SAY TIME". Now the
current timezone set on the system is "PST", and I have a request to
2007 Jun 24
3
Nokia N95 + Dial Plan
Hello All,
Recently I added some Nokia N95 customers and it worked pretty good.
Now the customers are complaining about the dialing rules...
They are used to dialing +12486543210 and +4479XXXXXX for long distance
calls.
Is there anyway to create a "+" sign dial plan which will allow them to
dial a number with "+" sign.
Cheers,
Nitesh
2007 Jun 27
2
.call file
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/1000 at from-internal
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data: recordvoice.php?id=3453&name=asterisk
Cheers,
Nitesh
2011 Feb 24
1
missing argument on AGI
Hi All,
I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
[callback-outbound]
exten => _00.,1,Macro(callout|${EXTEN})
[macro-callout]
exten => s,1,AGI(getchannel.php|${ARG1})
exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
exten => s,3,Hangup()
but for some reason i am not receiving the argument:
Executing [s at macro-callout:2]
2004 Jun 28
2
AGI->Exec Problem
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI->Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30");
I'm trying to record voicemail to the file name stored in $vmfile with
a silence timeout of 30. However, this is not being parse by AGI or
Asterisk correctly,
2004 Sep 12
1
Monitor and AGI - doesn't record much!
I have setup as per the monitor example configuration on the wiki site and
all works well for an extension dialing 8 then the number. However, if I
dial from an AGI script the recording stops after a few seconds. I see an
extra answer in the console and suspect that is the reason. Could any kind
soul help me to get around this?
Extensions.conf..
exten =>
2004 Jan 20
1
help - recording both sides of a conversati on
This is what I'm doing it gets you both sides of the phone call...small
size...and playable on windows through a share. My notes:
On redhat 9 I have to run the following command for asterisk to start
LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc
[macro-record-on]
exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1})
exten => s,2,Monitor(wav,${CALLFILENAME})
;exten =>
2005 Feb 14
2
Can't run AGI for outbound call
Hi
Just installed Asterisk on a Debian Woody/testing.
I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago).
The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory:
the test.call file looks like this:
#Simple test call script.
#call my
2008 Jan 15
3
Interrupt the swift text
Hi,
I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift text to audio.
My functionality is as for example i make this text to audio deliver the person called.
Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to cancel."
extension.conf dialplan:
[dialout]
exten =>
2007 Jan 08
1
No CDR from Outbound Call
I have a little call recording script that I am running and it works
fine, but I have one problem. I get CDR when a user calls into the
extension, but I do not get CDR for the call that it makes outbound on #
17. Any idea why? Here is the extensions info:
[default]
exten => 2211,1,Answer
exten => 2211,2,Wait(1)
exten => 2211,3,Playback(/etc/asterisk/recording/getshop)
exten =>