Displaying 20 results from an estimated 10000 matches similar to: "GETVARIABLE and IAX"
2012 Sep 06
1
Asterisk Test Suite error
Hi,
i am trying to install the Asterisk test suite on my ubuntu system , i have
followed all the installlation steps as mentioned in the link (
http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/) ,
but when i am trying to run the script some of the test cases are PASSED
and most of them are FAILED and SKIPPPED. So please help me out to do the
testing correctly.
The
2005 Jun 05
0
Re: Bison, Flex, Conditional Expression
To any that may be interested in the implementation of the conditional expression
in the expression parser (ast_expr2*) in asterisk, I've filed the patch at:
http://bugs.digium.com/view.php?id=4459
Right now, a comment has been added noting that the IF func provides this capability,
and asks if both would really be necessary. It's a good question. I haven't been
following the
2018 May 28
2
Dial to FastAGI application appears as 1-second CDR - how do I fix?
In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very
short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR
2006 Oct 23
0
SIP_HEADER function; what names are available?
Minor update - use the following:
> if (strcasecmp(data,
> "x-Asterisk-Request-URI-pseudo-header")==0)
> {
> ast_copy_string(buf, p->initreq.rlPart2, len);
> -----Original Message-----
> From: Steve Langstaff
> Sent: 23 October 2006 09:58
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users]
2007 Oct 12
0
Changes to 'refs/tags/0.5.3'
Tag '0.5.3' created by Benjamin Otte <otte at gnome.org> at 2007-10-12 17:13 -0700
release 0.5.3 ("Tennis")
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Version: GnuPG v1.4.6 (GNU/Linux)
iD8DBQBHD50fvMv5VCdLq8QRAj21AKC+kWUT9YN0gGDxLNS6Bequt5ufjgCgtYgd
9zH6AzP7/BHRfo0j1tw80V8=
=wOYr
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Changes since 0.5.2:
Benjamin Otte:
first try in getting movie
2007 Jul 07
0
Branch 'as' - 7 commits - libswfdec/swfdec_as_context.c libswfdec/swfdec_as_interpret.c libswfdec/swfdec_as_object.c libswfdec/swfdec_as_object.h libswfdec/swfdec_as_super.c libswfdec/swfdec_as_super.h libswfdec/swfdec_as_with.c test/trace
libswfdec/swfdec_as_context.c | 7 +-
libswfdec/swfdec_as_interpret.c | 35 ++++++-----
libswfdec/swfdec_as_object.c | 12 +++
libswfdec/swfdec_as_object.h | 5 -
libswfdec/swfdec_as_super.c | 46 +++++++++++++-
libswfdec/swfdec_as_super.h | 2
libswfdec/swfdec_as_with.c | 2
2005 Feb 03
1
FastAgi Help
Dear List
after a lot googling and watching source example of FastAGI i cant find a
simple way to convert a very simple perl AGI script... perhaps im not a
developer..
Why i have need to use FastAGI?...Very load CPU usage on my box... with only
100 calls..
So i have two way res_perl or FastAGI on some other box..
I cant test res_perl becasue when i try to compile it i have this error:
2010 Dec 22
0
CDR on MySQL
What would it do if you
exten => h,1,ResetCDR(w)
exten => h,2,NoCDR()
exten => h,3,DEADAGI(get-unqiueid.php)
I have not tried it but in theory it should write the first CDR and then
kill the write of the second NO ANSWER CDR.
Let me know if it works for you as I may need to do it on some of my h
exten code as well.
Bryant
----------------------------------------
From:
2007 May 17
4
FastAGI hangs up channel if server is not available
Hi all,
Running 1.2.14
When I call a FastAGI script such as this script for an incoming call:
[calldirect]
exten=>s,1,Answer()
exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)})
exten=>s,3,Goto(check_time,s,1)
and the FastAGI server is not running (Asterisk gets "connection
refused" TCP error), Asterisk just terminates the call like so:
May 17
2007 Jan 28
0
Trouble outgoing VOIP Provider Calls
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Extensions No Problems
Panasonic Ext -> SIP Extensions No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1]
2006 Apr 27
1
PrivacyManager & FastAGI: Rewrite or use?
I'm building an app that will do the following:
1. Force the caller to record their name.
2. Dial the party to call.
3. Play a short menu:
1 = Accept Call
2 = Decline Call, go to VM if available
3 = Accept Call forever, never ask again
4 = Decline Call forever, block number, get rid of caller
4. do things based on that choice.
I'm
2010 Nov 23
2
Function SIP_Header not registered
Hello,
I'm trying to use SIP_HEADER function on my dialplan but I receive this
message (on the console):
pbx.c:3367 ast_func_read: Function SIP_Header not registered
Why?
Thank's
- Bakko
2007 Feb 05
0
Help - Received response: "Forbidden" from'"Unknown"
I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect).
When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel
2007 Feb 04
1
Help - Received response: "Forbidden" from '"Unknown"
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Ext No Problems
Panasonic Ext -> SIP Ext No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",
2007 Dec 01
1
REFER mesage extraction using SIP_HEADER
Hi * users,
I would like to extract the information present in the SIP REFER
message that comes to asterisk. Would SIP_HEADER() allow me to do that
? I have used SIP_HEADER() for extracting the to and from SIP headers
previously.
Thanks
Regards
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel: 1-646-387-5998
2007 Dec 06
0
Perl FastAGI service port.
In the Perl FastAGI API, how does one set the port the service runs on?
[root at donkey queue_login_arbiter]# perl arbiter_agid.pl
2007/12/06-17:16:27 Evariste::QueueMemberArbiter (type
Asterisk::FastAGI) starting! pid(31737)
Port Not Defined. Defaulting to '20203'
Binding to TCP port 20203 on host *
Group Not Defined. Defaulting to EGID '0 10 6 4 3 2 1 0'
User Not Defined.
2013 Jan 29
1
Fast AGI library/support for C & C++
Dear All,
Is there anyone who is having FastAGI support for C & C++?
We do have FastAGI working for the JAVA and rest of the language / script.
But I am unable to find FastAGI for C/C++.
Please let us know how to write FastAGI using C/C++.
Thanks in Advance,
Kashyap
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2005 Jul 18
1
FastAgi ...fastagi-mapping missing error
salam .... if anyone uses fastagi then plz help me...
The AGISERVER starts successfully:
Jul 18, 2005 2:54:50 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Thread pool started.
Jul 18, 2005 2:54:51 AM net.sf.asterisk.util.impl.JavaLoggingLog info
INFO: Listening on *:4573.
but I m getting this error when i call extension
Jul 18, 2005 2:55:13 AM
2005 Aug 04
3
SIPPeersAction class file not found in the Asterisk-java.jar file
Hello Everybody,
I am working on Fastagi and I am making use of
Asterisk-java. But I don't find the class file for SIPPeersAction. Hence I
am getting the error message when compiling my java code.
----------------------------------------------------------------------------
------------------------------------------------
[root@localhost asterisk-java-0.1]#
2006 Oct 18
0
cut ip adress from caller id number display (ci$co 7941)
I'm playing with phone ci$co 7941 with sip image (8.02SR1),
strange is, that phone displays caller id number with ip address of
asterisk server like "8210@172.20.24.11"
I think, this is some bug in firmware, but I would like to find some
workaround,
maybe using SIP_HEADER function, but seems, that this can be used only
when calling from SIP to SIP,
i.e. not possible to use