Displaying 20 results from an estimated 30000 matches similar to: "CDR END TIME in correct in 1.8+"
2012 Jan 12
1
how to set callerid in php AGI file.
Hi,
I am using phpagi for agi scripting. and want to update callerid number but
didn't get any success. please help me how to update PHPAGI is new for me.
Below is the code which I write.
#!/usr/bin/php -q
<?php
set_time_limit(30);
//require(.phpagi.php.);
include("phpagi.php");
$agi = new AGI();
//answer the call
$agi-> answer();
2012 Sep 05
6
Async AGI
Hi,
Is there a way to execute next priority in the dialplan if you have called
agi:async? I want to play warning message if adhearsion is down. Currently
I wasn't able to make it work. The dialplan execution ends after the first
priority.
[incomming]
exten => _X.,1,AGI(agi:async)
exten => _X.,2,Answer
exten => _X.,3,Playback(some-message)
exten => _X.,4,Hangup
Regards,
Pavel
2013 Feb 15
6
Cisco 7942 Connected line ID
Hi,
Is it working for anyone?
I have tried with
trustrpid=yes
sendrpid=yes/pai
but can not get it working, Asterisk cli shows prevented message like this.
Connected line update to SIP/1231-00000200 prevented
Regards,
Zohair Raza
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2010 Sep 02
4
agi playback to execute say.conf settings
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number reading.
In the extension.conf:
--------------------------
[number-to-voice]
exten => 8765,1,playback(num:344345,say)
exten => 8765,n,hangup
It executes corresponding say.conf script and produces good results for me.
but when I write it in agi does not working. Here is agi debug output from
asterisk.
2013 Jan 02
3
Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,?)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..)
I have looked through all arguments of Dial
2012 Jan 03
4
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi,
Please help me understand the following applications and what are its
advantages if we compare between each of them.
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Regards,
Kaushal
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2012 Jul 13
8
How to set SIP to auto answer in the dial plan .
Hi,
I am trying to write dial plan for sip to auto answer (auto attend) the
incoming call to the sip phone.
- If i call from sip1 to sip2 then sip2 should automatically answer the
call and play some sound file.
I am trying to do this but as new to the asterisk dial plan configuration ,
so not able Todo this.
help me if anyone already done this setup.
Regards
Upendra.
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2010 Oct 29
2
Video based Asterisk Training
Hi Friends,
We have created a video based training for Asterisk in English and Urdu.
Please check them and let us know how we can improve them for no-voice
users.
http://www.youtube.com/watch?v=KXq9g8UiGnQ
http://www.youtube.com/watch?v=MID2RvgdD7s
http://www.youtube.com/watch?v=_LbDUdAGfSY
http://www.youtube.com/watch?v=J9Chkrg7E-M
http://www.youtube.com/watch?v=MsC12wc9ZnU
2012 Jan 04
2
asterisk -> AGI (perl) -> sqlplus (oracle)
Hi all,
I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
Currently my AGI is working fine in my two servers but not in my other four
servers. When I tried execute an AGI (as a user asterisk) in command line
it works fine (even I also declare environmental variables in user profile
and in my AGI), but when I tried to call my AGI (perl) in dial plan, it
don't get
2010 Sep 16
5
AGI Delimiter in 1.6
Hi
I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan and pass to an AGI script. I am now trying to set the
same thing up in 1.6 However because the argument delimter in 1.6 has
changed from pipe to comma this breaks as the
2010 Nov 03
1
Gotoif changed in 1.8?
Hi Gang,
I'm testing 1.8.0 on one of my machines and this snippet
"chokes" on line 7 (works fine with 1.4.30)
[tb-account-balance]
exten => s,1,Set(BALCOUNT=0)
exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} ))
exten => s,n(runagi),Set(TEST_RETURN="NONE")
exten =>
2011 Dec 28
1
cdr call time
Hi team,
On event of no answer in CDR the starttime and endtime of call remains the same.
Is there any way how can actually track call originate time and call end time.
Thanks
Vinod dharashive.
Sent from BlackBerry? on Airtel
2011 Dec 15
3
Play audio file for both Caller and Callee in a call
Dear all,
Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee?
A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor.
Thank you!
________________________________
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n
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
2013 Mar 06
1
Asterisk crashed
Hi,
I am running asterisk 1.8.14.0, It was running fine for last few days and
suddenly crashed today
In logs I can see that abrt tried to save the core dump but it couldn't
Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac
ip 0000000000533c19 sp 00007f7db9ce3af0 error 4 in asterisk[400000+1d1000]
Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528
2012 Jun 15
1
Does Asterisk support AMR and AMR-WB
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.
Now I have to setup a SIP server/SIP PBX in our Lab for test, does the
Asterisk
support these codecs and RFC4867 ? If no, there has any plugin to support
this ?
Also, any other Server/PBX which
2013 Jan 03
3
faxdetect on/off on the fly?
Hello,
We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
suggest a workaround?
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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2011 Apr 05
4
agi voicemail callback
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.
For instance, a caller leaves a voicemail, the voicemail will then call the
owner of the voicemailbox determined by a database look up.
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2010 Aug 27
1
asterisk-users Digest, Vol 73, Issue 58
On 8/27/10, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users at lists.digium.com
>
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2009 Jun 24
7
PHP AGI Not Working and Odd Behavior
Hi,
I'm running asterisk 1.4.22 on a debian server.
I have php5 installed and it works correctly command line.
When trying to run a php script via AGI, I get messages such as:
GI Tx >> I>
AGI Rx << #!/usr/bin/php5 -q
AGI Tx >> 510 Invalid or unknown command
The scripts are completely executable and owned by asterisk
-rwxr-xr-x 1 asterisk asterisk
Googling is not helping