I was tasked with developing an application that would be able to tell someone their cellphone credit balance via some IVR system. I dont know where to begin. Does anyone have a knowledge of GSM who could be able to give me some directions for this project.
You should look into Asterisk AGI scripts. They allow you to do more than Asterisk API. I currently have lots of AGI scripts in Perl but it says that Java is also supported. Maybe this link will help you. Basically you will create some extended Java app which will return you current Balance. Asterisk IVR would read this value from AGI and then play it.
Asterisk may be a good solution for your problem. Asterisk provide the API Asterisk Gateway Interface(AGI), that allows you to control the call flow using an external programming language, including Java.
Some of the libraries/frameworks that implement AGI in Java are: astivetoolkit.org and asterisk-java.org.
Good luck.
Related
I am using self developed CTI system and trying to integrate with softphone.
Currently I have integrated with AGEphone (https://www.ageet.com/home-en) however, the quality of the sound is pretty bad. The only reason why I am using AGEphone is it has protocol to hang up the call.
So I was wondering if there are any other softphone which has protocols to hang up from browser feature.
Doesn't matter if the software is free or paid.
Any recommendation would be appreciated.
You can hangup any call using asterisk AMI or ARI interface. No need special software for that.
Hi,
I'm busy developing a web interface for the asterisk PBX.
I'm looking for a way to initiate warm transfers via the web interface using the AMI.
I know that it's possible to initiate a warm transfer from the handset itself, but the requirement here is that it be done from the web interface.
I've done a fair amount of googling on the subject but I've not found anything thusfar.
Anybody know anything?
TIA.
To see all of the available manager commands, use the "manager show
commands" CLI command.
You can get more information about a manager command with the "manager
show command " CLI command in Asterisk.
https://wiki.asterisk.org/wiki/display/AST/AMI+Manager+Commands
You also can consult this page(note, that you still have check on your asterisk version like described above)
http://www.voip-info.org/wiki/view/Asterisk+manager+API
You are probably asking about Redirect command to some special context with dialplan support.
Does anyone out there have experience using BlueMix push notification invoked from code written & deployed on Z enterprise (Zos)? What were the basic coding & connectivity components you used?
It's easier than you think.
BlueMix Push has a REST API, so you can use any development tool capable of sending an HTTP request. In our case, we do it with a simple Java app that runs perfectly fine on z/OS, so long as you have network connectivity to the outside world. If you're not into Java, it could be done in C/C++ or even with a scripted utility like Curl or WGET.
Most sites will have some sort of HTTP Proxy and firewall, so you'll need to route your HTTP request through these layers, and this is a good reason to write the code in Java, since it's drop-dead simple. Oracle documents it here: http://docs.oracle.com/javase/tutorial/networking/urls/connecting.html, but you can also just search StackOverflow for plenty of examples.
I'm not sure this is even possible. I'm not a z/OS expert by any stretch but I'm guessing you won't find a suitable compiled binary for that architecture.
We use Asterisk and Hard-Phones. Now we want to have Instant messaging on our Windows 7.0 and 8.0 PCs. Also we want to give them the ability to transfer files and have video conference.
We want our admin be able to manage permissions on these features.
Is it possible to integrate these features with Asterisk? If yes, what solutions do you offer?
You can use sip message in asterisk, but it will be not so easy to maintain and debug.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MessageSend
Better use independed jabber message server on same host.
Video conference can be done via asterisk, but it is hard to setup and allow only one person speaking to be shown.
http://www.voip-info.org/wiki/view/Asterisk+video
Right Now I'm using Asterisk 11 on centos 6 operating system and using asterisk-java-1.0.0.m3.jar to establish communication between my java application and Asterisk 11.
I'm able to connect my java program with Asterisk
Issues I'm facing are
1.always at console message coming "Unsupported Version Asterisk Use at your own risk"
2.Unable to set property 'state interface' to 'SIP/0010' on org.asteriskjava.manager.event.QueueMemberStatusEvent:no setter found.Please report at http://jira.reucon.org/browse/AJ
My questions are:
1.Is there is asterisk-java-1.0.0.m3.jar source available anywhere so that i can edit source code and use it
2.I'm able to establish basic sip call,but getting issue as while making other type of calls which use other entities of same station like 0110 & 0210, which saved in extensions.conf & which work fine in Asterisk1.6 but not in Asterisk11.
As per my observation in Asterisk i added AppKonference module which is 3rd party module. & In java Program i write commands for that to establish channels using those entities.
But for communicate AppKonference in Asterisk API helps when Asterisk1.6 but not in 11.
Please suggest what I need to do is there is other asterisk and asterisk jar compatibility issue,if yes then what i need to do.
Please help
Sure, source can be found at
https://github.com/srt/asterisk-java
Also you can change agi compatability in /etc/asterisk/asterisk.conf