i am developing an application that shall allow users to access their mail boxes using their phones.
I developped an IVR menu of which users will make use to retrieve their mails. More specifically I write an AGI in perl to retrieve the mails. The issue is that the AGI execution takes quite a long time.To this end, I need to make my users hear some music played back in the background while the agi being executed. That is, i need to make the "AGI" and "MusicOnHold" applications in my dialplan execute simultaneously. Any clue how I can achieve this?
This may not be the answer you expected, but if you plan to create a proper application for doing stuff over the phone, have a look at other solutions apart from asterisk. Both yate and (as far as I know - never tried this one for anything serious) freeswitch implement a more asynchronous interface where you can connect an incoming call to whatever sound source you want without losing control over it. They may be a much better choice for advanced applications.
Asterisk is cool for simple logic, but if you need some processing in the background and leaving the call in some state for an undefined time, it becomes quite hard to manage.
You can issue SET MUSIC ON from you AGI applications, process your calculations, then issue SET MUSIC OFF.
If you want the music to be interruptable, issue this:
CONTROL STREAM FILE <filename> "0123456789*#" "0" "" "" ""
The second parameter is the list of digits able to interrupt the music.
It's better not to mix AGI and dialplan commands.
AGI is capable of everything dialplan is capable of, but AGI is implemented in a full fledged programming language and therefore gives you more control over the execution logic.
Is the complete execution taking a long time or just starting up the app? If the whole execution takes too long then you can do what Quassnoi says, start the music as soon as the AGI program starts.
If startup time is the problem, then you might need to implement something with FastAGI, so that your app is always running and listening for connections over a socket. The performance is much much better in an environment with a lot of AGI calls.
Related
I'm getting started with Asterisk and I have to develop an app that will only be used to call people (an alarm system).
I'm using Java and Asterisk-Java, and I read that on their GitHub:
Did you know you can originate directly into AGI? This is really powerful and eliminates the need for dialplan when doing AGI.
So, if I get it properly, that means I can do without a dialplan.
Though, I can't get Asterisk-Java to work properly, but I don't know it is because I have no dial plan or for some other reason.
Thanks
Yes, you need dialplan for anything except outgoing calling(initiated from asterisk side).
If you using only Originate and not need hangup handlers, caller id etc etc you can go without dialplan.
But in most cases it will be really tricky and usless.
Non-blocking sends/recvs return immediately in MPI and the operation is completed in the background. The only way I see that happening is that the current process/thread invokes/creates another process/thread and loads an image of the send/recv code into that and itself returns. Then this new process/thread completes this operation and sets a flag somewhere which the Wait/Test returns. Am I correct ?
There are two ways that progress can happen:
In a separate thread. This is usually an option in most MPI implementations (usually at configure/compile time). In this version, as you speculated, the MPI implementation has another thread that runs a separate progress engine. That thread manages all of the MPI messages and sending/receiving data. This way works well if you're not using all of the cores on your machine as it makes progress in the background without adding overhead to your other MPI calls.
Inside other MPI calls. This is the more common way of doing things and is the default for most implementations I believe. In this version, non-blocking calls are started when you initiate the call (MPI_I<something>) and are essentially added to an internal queue. Nothing (probably) happens on that call until you make another call to MPI later that actually does some blocking communication (or waits for the completion of previous non-blocking calls). When you enter that future MPI call, in addition to doing whatever you asked it to do, it will run the progress engine (the same thing that's running in a thread in version #1). Depending on what the MPI call that's supposed to be happening is doing, the progress engine may run for a while or may just run through once. For instance, if you called MPI_WAIT on an MPI_IRECV, you'll stay inside the progress engine until you receive the message that you're waiting for. If you are just doing an MPI_TEST, it might just cycle through the progress engine once and then jump back out.
More exotic methods. As Jeff mentions in his post, there are more exotic methods that depend on the hardware on which you're running. You may have a NIC that will do some magic for you in terms of moving your messages in the background or some other way to speed up your MPI calls. In general, these are very specific to the implementation and hardware on which you're running, so if you want to know more about them, you'll need to be more specific in your question.
All of this is specific to your implementation, but most of them work in some way similar to this.
Are you asking, if a separate thread for message processing is the only solution for non-blocking operations?
If so, the answer is no. I even think, many setups use a different strategy. Usually progress of the message processing is done during all MPI-Calls. I'd recommend you to have a look into this Blog entry by Jeff Squyres.
See the answer by Wesley Bland for a more complete answer.
i am setting up a DISA in asterisk .
what i would like to do is play a music to the caller while asterisk is receiving the DTMF.
the DTMF would stop playing when it receives DTMF for "#"
i have tried using background() and playback() command , but they do not serve the purpose!
any ideas?
khan
You can try using phpagi. There is a function called fastpass_get_data, where you can get dtmf input while playing a sound.
http://phpagi.sourceforge.net/phpagi22/api-docs/phpAGI/AGI.html#fastpass_get_data
I'm assuming just using the builtin DISA application doesn't meet your needs. If it does, just use that.
There are different ways to program asterisk. You can use normal dialplan, ael dialplan, AGI, AMI, even write your own modules in C.
You can do more things in dialplan, but often using a real language through AGI results in more readable code.
Using the dialplan applications StartMusicOnHold, StopMusicOnHold, and either Read or WaitExten, you should be able to do what you want. Run asterisk -rx 'core show application StartMusicOnHold', etc, to read about each of those.
You basically want to run StartMusicOnHold, then gather digits using Read or WaitExten until you're done (perhaps collecting them into a channel variable), then run StopMusicOnHold.
I'm playing a game and i'm trying to send some custom requests to the server in order to perform some tasks easier .. While i will gain little to none from this, i have become very interested in the educational part of it.
Since the game runs partially on client via a .jar and/or a .cab file i think it is run by JVM - correct me if im wrong
I have captured some traffic send by the game via wireshark. The protocol is TCP and it looks like this:
!, 1338,102,264,0.0 ,0.0,32433553,0, 102,264,
Nevermind all the numbers - thats for me to figure out.
But when i create and send a similar packet via a couple of different programs it always fails. This is of course because i am sending the wrong sequence number along with the TCP-packet.
So in order to not mess up the sequence-number i figure i will have to inject the process running the game and then somehow make it send my custom packets.
How do i go about that ?
You can't mess with the TCP sequence number in pure Java. Java doesn't even do that itself, the TCP stack does all that.
It is most unlikely that this is your real problem.
Part of this question is I'm not even sure what exactly I'll need to ask, so I'll start with the situation and work out from there.
One project I'm working on involves use of COMET via the aspComet library. The use case of the program is somewhat of a collaborative slideshow. One person runs the bulk of it, with one or more participants able to perform certain actions. Low latency between when an action is performed on screen
Previously, it was just running on one server. Now, we're wanting to scale it out a bit, more for reliability than performance reasons. So, we have some boxes out in Rackspace's cloud and all that fun stuff.
I knew from the start of this that I was going to need to make some changes to the way the COMET stuff works since different people in the same "show" might be on different servers, and I have no way of knowing what "show" they belong to until after they have already arrived on the site.
I initially tackled this using the WCF Mesh provider, which wasn't well documented to start with, now I'm running into issues with dispatching messages to it sometimes get lost, or delayed (I'm not 100% sure what is going on there), but it screws up the long poll for COMET and breaks things in rather strange ways (Clicking a button may trigger an event, or it'll hang for 10 seconds {long poll duration} and not actually do anything).
More research leads me to believe one of the .Net service bus providers may do what I need. However, I can't find examples that would cover what I need:
No single point of failure (outside of a database)
No hardcoding of peers.
Near-realtime (no polling, event based would be best)
My ideal solution would involve that when a server comes up, it lets other servers know of its existence (Even if it's just a row in a table somewhere), and they can start sending broadcast messages among each other, with each server being both a publisher and subscriber. This is what I somewhat had in the WCF Mesh provider, but I'm not overly confident in that code.
Can anyone point me in the right direction with this? Even proper terms to look for in the docs for service bus providers would be good at this point. Or are service buses not what I want? At this point I would settle for setting up a Jabber server on each web server and use that, if it could fit within my constraints.
I can't speak a ton to NServiceBus, but I expect the answers will be similar.
Single point of failure: MSMQ can use multicasting, which means each endpoint will broadcast it's existence and no DB table is needed. RabbitMQ uses this Exchange-to-Queue binding process which means as long as the Rabbit instance or cluster is up then messages still exist. RabbitMQ can be clustered, MSMQ cannot be. *Note: You might have issues with multicasting with Rackspace, no idea how they work. If so, you'll have to fall back on the runtime services for MSMQ (not RabbitMQ), that would create a single point of failure because everyone has a single point to coordinate control messages through.
Hard coding of peers: discussed above a bit; MSMQ's multicast handles it. Rabbit it can also be done, just bind queues to an exchange you want to listen to. MassTransit takes care of this for you.
Near-realtime: These both use messaging which is near real time. There's no polling in your message consumer code.
I think a service bus seems like a reasonable solution for what you're trying. Some more details would likely be needed, but the general messaging approach is correct. There are other more light weight messaging libraries if you decided you just want something on top of RabbitMQ and configure Rabbit to handle most of the stuff.
To get started with MassTransit, we have documentation up: http://readthedocs.org/projects/masstransit/ and mailing list http://groups.google.com/group/masstransit-discuss. Join the mailing list if you have future questions and someone will try and help you out.