Hi,
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.
What is your opinion
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.
Would
Olivier oza-4...@myamail.com 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
On Mon, May 18, 2009 at 10:02 AM, Scott Gifford
sgiff...@suspectclass.com wrote:
Olivier oza-4...@myamail.com 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
2009/5/18 Stefan Reuter stefan.reu...@reucon.com
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
2009/5/18 David Backeberg dbackeb...@gmail.com
On Mon, May 18, 2009 at 10:02 AM, Scott Gifford
sgiff...@suspectclass.com wrote:
Olivier oza-4...@myamail.com writes:
[...]
What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ?
I'm referring here to
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 use that
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)
exten = s,n,Hangup()
when the phone
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)
exten
在2009-03-19?06:53:56，Matt?Riddell?li...@venturevoip.com?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 19/03/2009 2:17 p.m., MaxGao wrote:
??2009-03-19?06:53:56??Matt?Riddell?li...@venturevoip.com???
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
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,Answer()
exten = s,n,Wait(10)
For a long time now we've used the astmanproxy process to handle manager
connections (75+ clients) so that these clients can tap into the
dialplan / send commands etc.
We use astmanproxy because at that time the manager connection routines
of asterisk did not cope well with numerous
Is there a way in the manager API to to tell it not to wait till the
first phone is answered before returning?
Jerry
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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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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 able to
restrict it based on IP address, and not require
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 able
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 would
Hi
Looks like it was it. Now it works OK. Thanks for help
Cheers
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Hi
I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial
out from manager's console and with Asterisk 1.4.X this settings were OK.
Action: Originate
Channel: SIP/384
Context: main
Exten: 102
Priority: 1
Callerid: 384
I could dial out, but with asterisk 1.6 I get this error.
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 29 Dec 2008 16:10:23 +0100
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Manager API
Hi
I have a problem with Asterisk-1.6.0.3-rc1
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
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));
2008/12/29 Andrew Nowrot andrew.now...@gmail.com
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);
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 andrew.now...@gmail.com
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
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 pretty
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).
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 as we're preparing for a pretty big upgrade at our call
center and I'm revamping all of the management apps I've written. I
Hi
What i need to do:
exten = 1001,1,AGI(Agent.agi)
agent.agi - login my interface in system
i would call to 1001 using Manager API and login interface in Asterisk.
This is possible?
Now i use originate. Something like that:
Action: originate
Channel: SIP/ekiga
Context: default
Exten:
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 to set a
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 the
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 cant set the
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 it recieves
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 using a simple
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
variable, asterisk responses with an error saying that the channel does
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
-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.
Does
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.
Does anyone know
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
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
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?
Thanks
Robert___
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Hi,
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.
?php
$strHost = 127.0.0.1;
$strUser =
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.
?php
$strHost = 127.0.0.1;
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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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 :Channel = SIP/0041435215301Context = defaultExten = 00982166501553Priority =
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 :
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
Hi,I'm having some issues with the manager api when it tries to redirect a call. If a call gets transferred to a person and the person doesn't answer, after the voicemail greeting the call gets dropped. As well when I try to redirect a call to a queue, there is only one way audio. If you need any
Hi,
I'm having a bit of trouble matching up Newchannel (and Newexten,
etc. etc.) events with the Originate that created them.
Basically, what I want to do is have software automatically initiate a
call, and then track the status of that call through to completion.
I can match to some
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 a
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... ;-)
rs Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Sunday, April 9, 2006 7:23:18 PMSubject: [Asterisk-Users] Manager API HelpHi 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 AP
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.
Thanks much.
Darren Ellis
Title: Manager API 'Redirect' is not working for both end of a call.
Hi everyone,
Here's what I come across. Phone A calls phone B through Asterisk and they are talking. An application uses manager api caught the channel IDs of both legs of the call. It then issue 'Redirect' with both
Title: Manager API mailing list
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.
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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
Is 1.2.0/1 still having problems with crashes due to having too many
connections to the manager api or has that been solved? If it is, does
anyone know roughly how many connections cause the crash or is it seemingly
random
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Does anyone have a sample on how to do a
supervised transfer via the Manager API.
Incoming Zap - SIP - xfer -
Zap
--Richard Cook[EMAIL PROTECTED]T: 705-223-2000
x2010
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Asterisk-Users
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
Hello,
I'm trying to write a php script that issues the QueueStatus and Queues
manager API commands to Asterisk and records the returned event data
into a MySQL table.
On voip-info.org, I see that QueueStatus returns such things as Queue,
Max, Calls, Holdtime, Completed, Abandoned,
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 5038.
Has anyone had experience using the Manager API with asterisk?I'm
using it to originate calls, and all seems fine and dandy.However, if I
set the "variable" parameter to be too long (longer than 245characters), I
get the following error message:Jul 14 18:35:41 WARNING[27769]:
manager.c:1211
I have some agi scripts that use the manager API. They just quit working
this afternoon.
It seems that asterisk quit responding on port 5038.
I can't even telnet to that port. (connection refused)
I had been making some changes to extension.conf but that's all.
I even went so far as to reboot
Hi,
I have a program that sends commands to the Manager
API and places outgoing calls.
I want to make it call a number, then pause, and dial
an office extension. Is this possible? How?
Action: Originate\r\nChannel: +
asteriskVoipChannel + / + PhoneNumber +
\r\nContext: + context +
Does the manager API have the option of showing timestamps of events?
I am trying to log events into a database and I need timestamps of when the
events actually occurred.
Is the time lag between events occurring and receiving them in the manager api
very low? I suppose it if is I could
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
Hi
Luca,
I
am trying to implement your solution for outbound calls through Manager API and
I get OutgoingSpoolFailed. Did it work for you? Can you give me more details
about your example?
Thanks
a lot,
Lior
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I am attempting to call DundiLookup from the manager API.
Action: Command
Command: dundilookup 401
I get back no such command.
I have CVS as of 3/31/05.
What am I missing so I can have the manager API return to me the
dundi lookup information?
Thanks,
jerry
I am using the manager API for "show channels".
If I have a multi line phone extenstions 510 - 515
and 510 has a call on hold and 511 has a call on hold
and I am answering 512 the manager API show channels
doesnt seem to tell me that 510 and 511 are on hold?
They are reported as Up.
How do I
I read the Wiki pages about the Redirect command, but,if I want to do a redirect into a MeetMe room, from a *remote* machine, how do I *query* Asterisk and get the Channel details?i.e the values for the Channel and ExtraChannel.I am using *SIP only*.Also, when redirected, one end Hangs up. Is this
_mailing_listSubject:
[Asterisk-Users] Manager API - Redirect command
I read the Wiki pages about the Redirect command, but,if I want to do
a redirect into a MeetMe room, from a *remote* machine, how do I *query*
Asterisk and get the Channel details?i.e the values for the Channel
and Ext
Hello,
I am trying to set a variable using the Manager API Setvar. I am
testing with a sample php code from the wiki. But when I run it I am
getting back the error:
ERROR:
Response: Error
Message: No such channel
Do channels have different names in the manager api than they do in
the
Good morning folks,
I am quite new to Asterisk but have successfully set it up
with some BRI lines, Cisco 7940/7960, queues, voicemail, XML
stuff and the flash operator panel. When playing with the
manager API to get some stuff integrated within our systems,
I stumbled across the Redirect command
and automated bank reconciliation features
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: February 04, 2005 7:41 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Manager API
Hi,
I try to use Manager API to originate a call
Asterisk 1.0.3 / TE410 / ISDN/PRI Zap channels
As I understand it using the Async: True in an
originate action is supposed do a Fast Originate
originate a call from a channel to an extension without waiting
for call to complete.
I'm finding no difference using Async or not, calls
always wait for
Hi,
I try to use Manager API to originate a call from a channel to an
existing extension. Based on sip channel show command, the Manager
initiates a call to the channel only. It doesn't generate a call to
the extension. So the originate call API of Manager is failed. I think
I pretty much
Asterisk 1.0.3
TDM400P/TE410P
Using originate()
call progress Events
normal progression
on completed call
Event: Newstate
State: Ringing
Event: NewState
State: up
On pri Zap channels call progress events
will wait @ State:Ringing until call FAILS
via timeout if
Hi All!
Let me explain the problem. When using the Originate
command from the manager api, the dialstatus variable returns results
for whichever phone picks up first, and in this case it is the IAX/2
connection. It doesn't matter if Zap/G2/XXX is set as the channel,
or an extension either.
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
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
?php
$fp = fsockopen(127.0.0.1, 5038, $errno, $errstr, 30);
if (!$fp) {
echo $errstr ($errno)br /\n;
} else {
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
?php
$fp = fsockopen(127.0.0.1, 5038, $errno, $errstr, 30);
if (!$fp) {
echo $errstr ($errno)br /\n;
} else {
: January 13, 2005 4:56 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Manager API !
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
?php
$fp = fsockopen
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 useful if
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
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't a solid
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
I'm not having any luck getting the ExtensionState action of the Manager
API to work.
The response is always success but the Status is always -1 which to me
means an error.
Here is a typical telnet session.
---
stockholm:~ # telnet localhost 5038
Trying ::1...
: [Asterisk-Users] manager API
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
Hi,
Where can I find a complete * manager api guide, the one one wiki is missing
informations like the monitor function for example,
Thnx
Serge
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Asterisk-Users@lists.digium.com
List - Non-Commercial Discussion
Subject: [Asterisk-Users] Manager API
Hi,
Where can I find a complete * manager api guide, the one one wiki is missing
informations like the monitor function for example,
Thnx
Serge
___
Asterisk-Users mailing list
Hi,
I'm playing with the agent/queue system. Everything work well with v1.0.3.
but I want the 'Action: Agents' in the manager API that is only on the CVS
version. So i switched to, but now the Queue/Agent system barely work. (my
agent don't get the call)
Where I can get a 'stable' CVS version?
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
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 the ALERT_INFO
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 to
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,
when a call is placed the local phone rings, once you pick
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 indicates.
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:
-
[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 Call Origination Variables
Well I tried just about every combination that I can think of as well
as
every
combination mentioned
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