Olivier wrote:
>> Would it help if you could use Asterisk-Java's implementation of the
>> Manager API for your script? Similar to what we already did for FastAGI
>> at
>>
>> http://blogs.reucon.com/asterisk-java/2009/05/13/scripting_support_for_fastagi.html
>>
>> If you could
2009/5/18 David Backeberg
> On Mon, May 18, 2009 at 10:02 AM, Scott Gifford
> wrote:
> > Olivier writes:
> >
> > [...]
> >
> >> What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ?
> >> I'm referring here to http://code.google.com/p/asterisk-php-api/.
> >
> > In my experience, asteri
2009/5/18 Stefan Reuter
> Olivier wrote:
> > I need a hack to query current calls coming in and going out an Asterisk
> > 1.6.1 system and send this list back as a custom UserEvent to other
> > listening endpoints.
> > For various reasons, I might need to write this hack in PHP though I'm
> > mor
On Mon, May 18, 2009 at 10:02 AM, Scott Gifford
wrote:
> Olivier writes:
>
> [...]
>
>> What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ?
>> I'm referring here to http://code.google.com/p/asterisk-php-api/.
>
> In my experience, asterisk-php-api works OK, but it's a bit slow. It
>
Olivier writes:
[...]
> What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ?
> I'm referring here to http://code.google.com/p/asterisk-php-api/.
In my experience, asterisk-php-api works OK, but it's a bit slow. It
determines when Asterisk has finished sending its responses by waitin
Olivier wrote:
> I need a hack to query current calls coming in and going out an Asterisk
> 1.6.1 system and send this list back as a custom UserEvent to other
> listening endpoints.
> For various reasons, I might need to write this hack in PHP though I'm
> more experienced with Asterisk Java.
Wou
oh, i am sorry, plain text :
>> hi, all
>>
>> asterisk 1.4.24 , zaptel 1.4.10.1 , E1
>>
>> Manager API Action :
>>
>> Action: Originate
>> Channel: ZAP/G1/888
>> Callerid: 12345678
>> Context: callout
>> Exten: s
>> Priority: 1
>>
>> extensions.conf
>>
>> [callout]
>> exten => s,1,
On 19/03/2009 2:17 p.m., MaxGao wrote:
??2009-03-19?06:53:56??"Matt?Riddell"
On?18/03/2009?9:58?p.m.,?MaxGao?wrote:
?hi,?all
??asterisk?1.4.24?,?zaptel?1.4.10.1?,?E1
??Manager?API?Action?:
?Action:?Originate
?Channel:?ZAP/G1/888
?Callerid:?12345678
?Context:?callout
在2009-03-19?06:53:56，"Matt?Riddell"??写道：
>On?18/03/2009?9:58?p.m.,?MaxGao?wrote:
>>?hi,?all
>>
>>??asterisk?1.4.24?,?zaptel?1.4.10.1?,?E1
>>
>>??Manager?API?Action?:
>>
>>?Action:?Originate
>>?Channel:?ZAP/G1/888
>>?Callerid:?12345678
>>?Context:?callout
>>?Exten:?s
>>?Priority:?1
On 18/03/2009 9:58 p.m., MaxGao wrote:
hi, all
asterisk 1.4.24 , zaptel 1.4.10.1 , E1
Manager API Action :
Action: Originate
Channel: ZAP/G1/888
Callerid: 12345678
Context: callout
Exten: s
Priority: 1
extensions.conf
[callout]
exten => s,1,Answer()
exten => s,n,Wait(10)
exte
Jerry Geis wrote:
> Is there a way in the manager API to to tell it not to wait till the
> first phone is answered before returning?
>
> Jerry
>
I found the Async: yes option.
Jerry
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On Thu, Jan 29, 2009 at 05:55:39PM -0600, Brooks Bridges wrote:
> I've been searching around for a while, and haven't found an answer to
> this question, so here goes:
>
> Does anyone know if AMI can be configured to allow requests from another
> client without having to authenticate first? I w
On 30/01/2009 12:55 p.m., Brooks Bridges wrote:
> I've been searching around for a while, and haven't found an answer to
> this question, so here goes:
>
> Does anyone know if AMI can be configured to allow requests from another
> client without having to authenticate first? I would like to be
Hi
Looks like it was it. Now it works OK. Thanks for help
Cheers
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2008/12/29 Andrew Nowrot
> I did not need to change the code. My manager.c already has all the lines
> you specified that are wrong.
>
>
>> did you re compile and re installed?
>> make
>> make install
>> after the code change?
>>
>> david
>>
>>
> Cheers
>
> ___
I did not need to change the code. My manager.c already has all the lines
you specified that are wrong.
> did you re compile and re installed?
> make
> make install
> after the code change?
>
> david
>
>
Cheers
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2008/12/29 Andrew Nowrot
> Hi
>
> Thanks for so fast reply, but I already have this part like this:
>
>
> static int action_timeout(struct mansession *s, const struct message *m)
> {
> struct ast_channel *c;
> const char *name = astman_get_header(m, "Channel");
> int timeo
Hi
Thanks for so fast reply, but I already have this part like this:
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
int timeout = atoi(astman_get_header(m, "Timeout"));
hi
that is a bug in manager.c
where saysstatic int action_timeout(struct mansession *s, const struct
message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
int timeout = atoi(astman_get_header(m, "Timeout"));
if (!ast_strlen_zero(n
I had the same problem, I think with that version. You need to use a more
current rc version. I think I am using rc2 but rc3 has been released as I
recall.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Andrew Nowrot
Reply-To: Asterisk Users Mailing List - Non-Com
> Isn't there one already?
Yeah, but none of them have worked for me...maybe their way of doing things
is just different from my approach but I wasn't happy with any of the
existing classes. I wasn't planning on releasing my code to the wild (I'm
not a programmer by trade I just play one on TV).
On Mon, Dec 22, 2008 at 09:04:13AM -0600, Wesley Haut wrote:
> Hi all,
>
> I know I'm probably stirring up a hornet's nest with this question/comment
> but I've spent the last few days working on a PHP-based class for the
> manager interface
Isn't there one already?
> as we're preparing for a p
In article <[EMAIL PROTECTED]>,
Rizwan Hisham <[EMAIL PROTECTED]> wrote:
> Can anybody help in parsing the manager events efficiently? Any ideas?
Since you're using perl, have a look at POE-Component-Client-Asterisk-Manager
here: http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager
Can anybody help in parsing the manager events efficiently? Any ideas?
On Fri, May 9, 2008 at 5:07 PM, Gunārs Grundāns <
[EMAIL PROTECTED]> wrote:
>
>
> On 5/8/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>>
>> On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
>> > Hi all,
>> > I am
On 5/8/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
> > Hi all,
> > I am using a simple perl script to connect with ast manager api. the
> script
> > tries to set a channel variable. It extracts the channel name from the
> > events
Thanx a lot.that was it. will never forget to remove the new
character again. Now its working fine.
On Fri, May 9, 2008 at 4:31 PM, Tony Mountifield <[EMAIL PROTECTED]>
wrote:
> In article <[EMAIL PROTECTED]>,
> Rizwan Hisham <[EMAIL PROTECTED]> wrote:
> >
> > same is the case in 1.6, i c
In article <[EMAIL PROTECTED]>,
Rizwan Hisham <[EMAIL PROTECTED]> wrote:
>
> same is the case in 1.6, i cant set the variable still.
My guess would be that you have a problem with line endings.
All lines received from the manager interface are terminated with \r\n,
not just \n. If you only strip
same is the case in 1.6, i cant set the variable still.
On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:
> On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
> > Hi all,
> > I am using a simple perl script to connect with ast manager api. the
> script
> > tries
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
> Hi all,
> I am using a simple perl script to connect with ast manager api. the script
> tries to set a channel variable. It extracts the channel name from the
> events it recieves after dial command. When i try to set the channel
> va
Thanks all, problem solved.
> Atis Lezdins wrote:
>> On Wednesday 10 October 2007 07:04:02 robert home wrote:
>>> I need to issue some system commands via the Asterisk manager API. From
>>> the
>>> CLI the ! (system command) works fine, but when connected via the
>>> manager
>>> API it fails.
>>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Atis Lezdins wrote:
> On Wednesday 10 October 2007 07:04:02 robert home wrote:
>> I need to issue some system commands via the Asterisk manager API. From the
>> CLI the ! (system command) works fine, but when connected via the manager
>> API it fails.
Yes - use the manager API to do an Originate by setting variable $CMD to
the shell code you want to execute, and then call a piece of dialplan
where the shellout will be executed through the System( $CMD ) command.
Note that this would enable an attacker to execute arbitrary commands with
On Wednesday 10 October 2007 07:04:02 robert home wrote:
> I need to issue some system commands via the Asterisk manager API. From the
> CLI the ! (system command) works fine, but when connected via the manager
> API it fails.
>
> Does anyone know why, or of a work around?
I believe, it's because
On Sat, 5 May 2007, Arun Kumar wrote:
Hi,
Is there any way that I can store my manager API output that is:
Read The Fine WiKi!!!
http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+PHP
Gordon
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Arun Kumar wrote:
> Is there any way that I can store my manager API output that is:
> My question is that is there any why using that I can get the QueueStatus
> and store the result in some text file for further processing.
>
>
> $strHost = "127.0.0.1";
>
If I understood your question correctly, you just need to reverse everything.
Channel = OUTGOING TRUNK i.e. ZAP/00982166501553
Context = default
Exten = internal extension that points to -> 0041435215301
Priority = 1
CallerID = 0041435215301
This will first initiate the call to the number 004143
On Wed, 2006-11-01 at 03:23 -0800, Ehsan Khosrowshahi wrote:
> Hi all,
>
> How can i originate a call from someone outside my sip-network (for
> example my PSTN home number) to one of my SIP number?
>
> I can originate a call from my SIP-network using this parameters in
> Originate call command :
This is the code I wrote to do that job, without getting into the full phpagi:function readManagerResponse($resp, $lookFor) { $value_start = strpos($resp, $lookFor) + strlen($lookFor); $value_stop = strpos($resp, "\r\n", $value_start); if (strpos($resp, $lookFor) === FALSE ){ ech
you could use the "Action: Command" action and pass the cli commands you need, like database show, database put, database deltree, etc...
or the DBput, DBget, DBdel manager actions...Obviously then you have to parse the answers...
Anyway, "show manager commands" on the cli is your friend... ;-)
Darren Ellis wrote:
> Hi All,
>
> Could someone send me a code frag on how to get a record from the
> asterisk database into a PHP variable via the Manager API?
>
> I can issue calls, etc. from Manager. But I'm not comprehending how to
> manipulate database variables.
Google for phpagi, it is
Wai Wu wrote:
> Hi all,
>
> I am new to this list. I have been looking for a Manager API mailing =
> list for a while, but could'nt find any. Is there a such list? Thnx.
Not at the moment.
There is an AstManProxy list for the manager proxy, but not really a
manager list as such.
--
Cheers
dont know how many, but i guess is better to use a ManagerProxy. In
voip-info.org you can find one written in C, and i guess is
multithreaded. I have never used it, i have my own manager proxy in PHP
for my own purposes.
best regardsOn 12/22/05, rushowr <[EMAIL PROTECTED]> wrote:
Is 1.2.0/1 still
> Does anyone know what the descriptions are for the data that
> "QueueStatus" and "Queues" manager API commands return? Any information
> would be helpful. Thanks in advance.
anything that I know about the events is in the javadocs of
Asterisk-Java. Have a look at
http://asterisk-java.sourcefo
output of netstat -lnp | grep asterisk does show the 5038 port?
post your manager.conf
best regards
On 7/13/05, Malcolm Bader <[EMAIL PROTECTED]> wrote:
> I have some agi scripts that use the manager API. They just quit working
> this afternoon.
> It seems that asterisk quit responding on port 50
i think the time between sent event from Asterisk and catch the event
with some other application is not important for most applications, so
you may save the timestamp from your own application.
And of course you have other option, modify the function:
int manager_event(int category, char *event,
You
should be able to get the full channel values by doing a "Action:
Command Command: Show Channels" and
picking your SIP extension out of the list it gives you of active channels. Then
you can take that and the channel that you are currently connected to, also
taken from the "Show
<< Has anyone tried the API before?>>
Of course. Here is a snippet taken from Nicolas Gudino's op_server.pl to
originate a call.
my $comando = "Action: Originate\r\n";
$comando .= "Channel: $originate\r\n";
$comando .= "Exten: $canal\r\n";
$coman
Thanks to all ,
I have now managed to get the ExtentionState to return a Status value
BUT.
It seems to always return -1 whether the phone is on a call or not. Am I
missing something ? I would have thought it should return some other
value when the line is engaged OR am I looking at completely the
On Thu, Jan 13, 2005 at 04:56:05PM -, Simon wrote:
> Hello all
>
> Has anyone had any success with the Manager API ?
>
> I am trying to check an extension status without too much luck I have
> the following
You can supply an ActionID in your request to track the response. This
is most usefu
<< Has anyone had any success with the Manager API ?>>
Yes, lots of success. I think we found that tracking the status of an
extension is the most reliable way to monitor extension state. We do issue
a PeerStatus request at the beginning but that's pretty time and resource
intensive. Since the
Simon wrote:
Hello all
Has anyone had any success with the Manager API ?
I am trying to check an extension status without too much luck I have
the following
$fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30);
if (!$fp) {
echo "$errstr ($errno)\n";
} else {
Guys,
After connecting to the * manager, each and every event is sent to the
connected client, right?
This means that if I install a client on each PC for monitoring incoming
calls, or pretty much anything else, it will create a lot of excess
traffic on my LAN.
Can I connect to the manager and tell
Hello,
You are correct, it's a lot of data for each client to parse through and a
lot of data for the se5rver to be sending out. It would just be easier to
use an AGI to trigger an action on the client computer, or you could just
use astGUIclient which already does what you are trying to do:
http
First class service, thanks a lot :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: mardi 4 janvier 2005 03:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Manager API
There really isn
There really isn't a solid description of every Manager API function, even
in the source code. And some of the features listed may not work the way you
think they should.
As for Monitor, there really isn't much more than to say that you send it a
channel and optionally a filename and it will start
Peter Osborne wrote:
Hi all,
I am using the Asterisk Manager API to originate calls and it is working well,
when a call is placed the local phone rings, once you pick it up you can here
the call ringing the other end. Now, I am using Polycom IP 300 and I have
them setup to auto-answer if I set t
ions Engineer
Akiva Corporation
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Peter Osborne
> Sent: Monday, November 15, 2004 2:09 PM
> To: [EMAIL PROTECTED]
> Cc: Peter Svensson
> Subject: Re: [Asterisk-Users] Manager API
Well I tried just about every combination that I can think of as well as every
combination mentioned and it still doesn't work. Not sure why, maybe it's
just not possible from the Manager API.
Pete
On Monday 15 November 2004 04:56, Peter Svensson wrote:
> On Mon, 15 Nov 2004, Brian West wrote:
On Mon, 15 Nov 2004, Brian West wrote:
> Ok to cut confusion here
>
> Its:
> Variable: _ALERT_INFO
> Value: somevalue
>
> Its always var/val via manager.
Not in the Originate action it isn't. This is what both the help
show manager command originate
say and what reading the source indicate
Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Manager API Call Origination & Variables
>
> On Mon, 15 Nov 2004, Peter Osborne wrote:
>
> > I am using the Asterisk Manager API to originate calls and it is working
> well,
> &g
On Mon, 15 Nov 2004, Peter Osborne wrote:
> I am using the Asterisk Manager API to originate calls and it is working
> well,
> when a call is placed the local phone rings, once you pick it up you can here
> the call ringing the other end. Now, I am using Polycom IP 300 and I have
> them setup
Hello Nicolas, first of all I want to thank you. You are the first guy
give me an answer. I already posted this issue two times but nobody was
interested in it.
I tried your sugggestion but it doesn't work. In the mean time I
upgraded to v1.0.2 but things remain the same or even worse ( I have
Hello,
Comments inline..
> The question is how to correctly handle failed calls.
> In my application I want to make hundreds of outgoing calls automatically.
> When the callee pick up the phone he gets a playback message and give an
> acknowledge by means of dtmf code.
> I make use of man
Hi,
On Thu, 21 Oct 2004 11:37:27 +0100, Ben Merrills <[EMAIL PROTECTED]> wrote:
>
> Is there a way to find out which agents are logged in, without waiting for
> the Agentcallbacklogin/off event?
>
> I'd like to be able to get the login status of all Agents when I connect to
> the Manager API.
Michael Devenijn wrote:
Everythings works great with asterisk exept one feature with redirect
: it doesn't redirect when ringing ...
Have you used astman with a new CVS? It works for me...
If not, you'll need to post more information for the list to help you.
Nick
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