Displaying 20 results from an estimated 20000 matches similar to: "analog lines running agi on hangup question"
2008 Aug 07
1
outgoing call file and agi detect busy
I am using asterisk 1.4.21 with outgoing call files.
I am call a line that is busy as you can see below.
How can my AGI ask what the status of the last call was
so I can tell if there was NO ANSWER or it was BUSY?
Thanks,
Jerry
-- Attempting call on SIP/401 for
smvoice_callprogress at smvoice-dialout:1 (Retry 1)
-- Got SIP response 486 "Busy" back from 192.168.1.161
2011 May 17
1
Question on AMI
I am using asterisk 1.4.41 and the AMI
I am trying to execute a command over AMI, specifically "core show
channels concise"
"sometimes" I get this back:
asterisk_command_show_channels() execute failed. 'Response: Follows[CR
][LF ]Privilege: Command[CR ][LF
]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF
]'
I'm not
2008 Sep 26
2
server and 2 uniden phones no ringing
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the canreinvite for no to yes but that made no
difference after restarting.
Very simple network. server, linksys router and 2 phones.
2007 Apr 19
1
Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine.
However, using outgoing call files the CS1000 is hanging up after I answer the call.
I dont know why?
Thanks, for any assistance.
Jerry
my sip.conf entry is:
[Nortel]
type=friend
dtmfmode=rfc2833
username=XXXXXXXXX
disallow=all
allow=ulaw
allow=alaw
2009 Oct 04
3
After call into console/dsp hangup hear ringing
I am running asterisk 1.4.26.1 and using ALSA not oss
dahdi 2.2.0
and libpri-1.4.10
I am calling into console/dsp I hear the audio just fine then after the
hangup I hear ringing
on the console/dsp.
Why would that be?
I found this bug for OSS https://issues.asterisk.org/view.php?id=13686
Does the same thing exist in ALSA???
some traces below
Jerry
== Parsing
2011 May 04
1
asterisk 1.4.35 to 1.4.41
Under 1.4.35 I get this message printed MANY times
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000 (g722)(4096)
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000
2004 Apr 16
1
Transfer through AGI
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="+1">Hi,<br>
<br>
I have the 4 port TDM card.
2008 Jun 30
0
how to have an agi check for dial tone on analog lines before dialing
hi,
I have an AGI running after an outgoing call file starts it up.
Everything works fine except if my line has a problem.
Trying to simulate this I unplug the line. So there is no dialtone.
How do I detect this and let the AGI know so I can try line 2, 3, 4 etc...
Detecting the the AGI or some other way is fine. I just need to know.
I am using a TDM804B card at this time.
Jerry
2006 Mar 06
3
call manager integration
I am getting this error from call manager (4.0) and asterisk 1.2.4
I have canreinvite=yes on the call manager setup.
I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I see
this entry from call manager...
What might be the problem with my setup?
THanks,
JErry
----------------
<Date>03/06/2006
2005 Mar 07
1
Exec AGI after hangup.
Hi everybody,
I'm trying to implement a enhanced blacklist system using AGI and Perl, configuration in extension.conf is:
exten =>_numbera,1,AGI,blacklist_2_in.agi
exten =>_numbera,2,Answer
exten =>_numbera,3,AGI,xisco_1.agi
exten =>_numbera,4,AGI,blacklist_2_out.agi
The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in
2005 Sep 23
2
Execute php agi after channel hangup
Hi,
following I would like to implement:
1. I receive a call.
2. I hang up the call.
3. I execute a macro
I thought about using call files first... but they don't support macros, or?
Then I figured I could use php agi after I receive the call, hang up the
call with php agi and execute the macro with "exec ...".
Unfortunately, the php agi seems to die with the hangup. Does
2005 Jan 11
0
AGI Application Hangup when using AGI->getdata
Before coming in here , I had a deep dig into Google and couldn't find
an answer,
Simply spoken, using agi->getdat in an AGI application , the call
disconnects if digits are entered fast by user.
I'm certain that others have been though this problem, please pour your
experience here :-) .
Ali Mughrabi
2013 Sep 13
2
executing the h extension at the real hangup of the call
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always over SIP) I want to keep track of who answered and of the length of the call.
[outgoing-dev2]
exten => h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten => _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten => _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =>
2006 Nov 02
1
AGI Problems
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten => _X.,1,Set(TIMEOUT(absolute)=0)
exten => _X.,2,NoOp(${EXTEN})
exten => _X.,3,DEADAGI(live-full.php)
exten => _X.,4,Wait,2
exten => _X.,5,Hangup
The script is using phpagi-2 from http://phpagi.sourceforge.net/ and
2011 Apr 09
1
Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?
Hi Everyone,
Trying to run a php script after DeadAGI for A2Billing does it's magic. This
is the dialplan:
[a2billing]
exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN}
${UNIQUEID})
exten => _X.,n,AGI(a2billing.php,1)
exten => _X.,n,Hangup()
*exten => h,1,Wait(5)*
*exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})*
As you can see above, I even
2018 Jan 18
2
Handling a long-running agi on hangup-handler?
I know that hangup handlers (
https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish
quickly.
So it's no surprise that my speech to text agi which takes 8 seconds gets
killed.
However, can anyone think of a way round this? So, once the caller has hung
up, I need to take one of the channel variables, and pass it to a python
agi script which then does speech to text.
2013 Mar 07
2
Recording with MixMonitor and AGI
Hi,
I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan:
[macro-ccdev2-rec]
exten => s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten => _X.,1,NoOp(Will send call to ${EXTEN})
exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})
2009 Feb 13
2
Continue processing AGI script after hangup
All;
I wrote a PERL AGI script that prompts a caller to leave a message using
print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n";
When the caller is done, they need to press the # key. The message is then delivered.
However, the message is not delivered if the caller simply hangs up when finished.
If the user hangs up, the script ends right then. How do I keep on processing the
2008 Oct 16
2
DAHDI and wait 'w'
-- Attempting call on DAHDI/1wwwwww for
smvoice_callprogress at smvoice-dialout:1 (Retry 1)
[Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request:
Unknown option 'w' in '1wwwwww'
[Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec:
Unable to enable echo cancellation on channel 1 (No such device)
Does DAHDI not know about the W ??? I think zaptel used
2011 May 19
1
SIP 603 Declined after AGI execution
Hello everyone.
I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small
wholesale operation, so I configured A2Billing for not to answer the
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer
has it's own context, in which I set the following:
;=====in extensions.conf======