im have problem in ubluetooth module.
Standard in the characteristics to write is: FLAG_WRITE (default with reposnse) how to call with without reponse?
(gatt server)
Related
I've been using a sim900 module to replicate many of the functions found in a basic cellphones for an embedded project. I've been successful with most functions with the exception of entering options during a voice call. I am actually looking for a generic solution (e.g. GSM 07.07 etc.) although the GSM/GPRS Module I'm using is the sim900.
Scenario: I initiate a call using ATD<number>; ,then automated voice asks me to dial "1" for an option. How do I send the "1"?
I've search high and low for an answer. I've been through the AT command manual over and over again. Please help.
Very good start in using the official GSM specification, although I want to note that 07.07 has been superseded by 27.007 a very long time ago, and you should use that document (or 27.005 if relevant).
I initially expected there to be two ways of achieving this, dial string modifiers or DTMF command, but I looking up the dial string in the 27.007 specification I do not find the p (pause) modifier1 I was expecting, and nearest thing, the W (wait) modifier is ignored and only included for compatibility.
Sending 0 through 9, A through D, * and # during a call is done using DTMF, although in a GSM network this is signalled separately out of band rather than sending in-band analogue tones. There is a specific command for sending DTMF tones called AT+VTS (with a horrible syntax). So that command is the answer to you question. Invoke it either from online command mode or from another serial connection.
1 The reason I was expecting a p modifier to exist is that I am able to enter one in phone book entries, e.g. `"12345678p123" which will dial 12345678, wait for the connection to be established and then send 123 as DTMF tones. But this is then obviously something (only) the user interface voice call handler manages and not the AT command handler.
I am looking into the Swift Vapor framework.
I am trying to create a controller class that maps data obtained on an SSL link to a third party system (an Asterisk PBX server..) into a response body that is sent over some time down to the client.
So I need to send received text lines (obtained separately on the SSL connection) as they get in, without waiting for a 'complete response' to be constructed.
Seeing this example:
return Response(status: .ok) { chunker in
for name in ["joe\n", "pam\n", "cheryl\n"] {
sleep(1)
try chunker.send(name)
}
try chunker.close()
}
I thought it might be the way to go.
But what I see connecting to the Vapor server is that the REST call waits for the loop to complete, before the three lines are received as result.
How can I obtain to have try chunker.send(name) send it's characters back the client without first waiting for the loop to complete?
In the real code the controller method can potentially keep an HTTP connection to the client open for a long time, sending Asterisk activity data to the client as soon as it is obtained. So each .send(name) should actually pass immediately data to the client, not waiting for the final .close() call.
Adding a try chunker.flush() did not produce any better result..
HTTP requests aren't really designed to work like that. Different browsers and clients will function differently depending on their implementations.
For instance, if you connect with telnet to the chunker example you pasted, you will see the data is sent every second. But Safari on the other hand will wait for the entire response before displaying.
If you want to send chunked data like this reliably, you should use a protocol like WebSockets that is designed for it.
I have added International Domain Name support to an XMPP client as specified in RFC 6122. In the RFC it states:
Although XMPP applications do not communicate the output of the
ToASCII operation (called an "ACE label") over the wire, it MUST be
possible to apply that operation without failing to each
internationalized label.
However, with the domain I have available for testing (running Prosody 0.9.4; working on getting feedback from someone else about how Ejabberd handles this), sending a Unicode name in the "to" field of an XMPP stanza causes them to immediately return an XMPP error stanza and terminate the stream. If I apply the toASCII operation before sending the stanza, the connection succeedes, and I can begin authentication with the server.
So sending:
<somestanza to="éxample.net"/>
Would cause an error, while:
<somestanza to="xn--xample-9ua.net"/>
works fine.
Is it correct to send the ASCII representation (ACE label) of the domain like this? If so, what does the spec mean when it says that "XMPP applications do not communicate the output of the ToASCII operation ... over the wire"? If not, how can I ensure compatibility with misbehaving servers?
I need tips to build an infrastructe to send 1000 simultaneous voice calls (automated IVR calls with voicexml). Up to now i used asterisk with voiceglue but now i have performance issues.
The infrasturcture was like this:
the asterisk pulls request from queue
the queue consumer create a call file
when the call ends, call file is read and status is sent to the application server
To be honest, i am asking for tips to implement an infrastructure like callfire[1] or voxeo[2]?
[1]https://www.callfire.com/
[2]http://voxeo.com/
you can go with voxeo prophecy (http://voxeo.com/prophecy/) one of the good server which have the capability of making simultaneous voice calls
Note: The requirement which your are expecting to do will not only possible with voxeo prophecy it should also depend the web server like Tomcat, IIS e.t.c in case if you dealing with databases like Sql , Oracle e.t.c
Please do refer to know the architecture
http://www.alpensoftware.com/define_VoiceOverview.html
CallFire's API has a CreateBroadcast method where you can throw up an IVR using their XML in seconds. You can read up on the documentation here:
https://www.callfire.com/api-documentation/rest/version/1.1#!/broadcast
CallFire also offers a PHP-SDK, hosted on Github, with examples of how to do this. The SDK is minimal setup and allows you to easily tap into the APIs robust functionality. Version 1.1 can be found here, with instructions on how to get started: https://github.com/CallFire/CallFire-PHP-SDK
The method call might look something like this. Note the required dependencies.
<?php
use CallFire\Api\Rest\Request;
use CallFire\Api\Rest\Response;
require 'vendor/autoload.php';
$dialplan = <<<DIALPLAN
<dialplan><play type="tts">Congratulations! You have successfully configured a CallFire I V R.</play></dialplan>
DIALPLAN;
$client = CallFire\Api\Client::Rest("<api-login>", "<api-password>", "Broadcast");
$request = new Request\CreateBroadcast;
$request->setName('My CallFire Broadcast');
$request->setType('IVR');
$request->setFrom('15551231234'); // A valid Caller ID number
$request->setDialplanXml($dialplan);
$response = $client->CreateBroadcast($request);
$result = $client::response($response);
if($result instanceof Response\ResourceReference) {
// Success
}
You can read this:
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Main tip: you WILL have ALOT of issues. If you are not expert with at least 5 years development experience with asterisk, you have use already developed dialling cores or hire guru. There are no opensource core that can do more then 300 calls on single server.
You can't do 1000 calls on single asterisk in app developed by "just nice developer". It will just not work.
Task of create dialling core for 1000 calls is "rocket science" type task. It require very special dialling core, very special server/server tunning and very specialized dialler with pre-planning.
1000 calls will result 23Mbit to 80Mbit bandwidth usage with SMALL packets, even that single fact can result you be banned on your hosting and require linux network stack be tunned.
You can use ICTBroadcast REST API to integerate your application with reknown autodialer , please visit following link for more detail
http://www.ictbroadcast.com/news/using-rest-api-integerate-ictbroadcast--third-party-application-autodialer
ICTBroadcast is based on asterisk communication engine
I've already done this for phone validation and for phone message broadcasting using Asterisk and Freeswitch. I would go with Freeswitch and xmlrpc:
https://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
I'm new to Applescript, but I followed this handy tutorial, the first Google result for "Applescript webservice": http://developer.apple.com/internet/applescript/applescripttoperl.html
So I have (hopefully) a perl script giving me a webservice at http://localhost:8001
but when I run
set p to {"http://www.perl.com/pace/perlnews.rdf", "http://www.perl.com/pace/perlnews.rdf"}
using terms from application "http://www.apple.com/placebo"
tell application "http://localhost:8001" to return call soap {method name:"fetch_headlines", method namespace uri:"http://localhost:8001", SOAPAction:("http://localhost:8001" & "#" & "fetch_headlines"), parameters:p}
end using terms from
I get
"got an error: Transport error"
Google is failing me. Any ideas where to look?
Transport error mean that the communication endpoint does not respond. Do you forget start the perl soap-server script (for listening at port 8001)?
And after you start your perl SOAP server, your applescript SOAP-client code should be
set p to "http://search.cpan.org/uploads.rdf"
using terms from application "http://www.apple.com/placebo"
tell application "http://localhost:8001/Server" to return call soap {method name:"fetch_headlines", method namespace uri:"http://localhost:8001/Server", SOAPAction:("http://localhost:8001/Server" & "#" & "fetch_headlines"), parameters:{uri:p}}
end using terms from
ps: http://www.perl.com/pace/perlnews.rdf does not exists...