How-to use Asterisk box to dial an extension for testing

The Zoiper softphone will open an URL upon receiving a call on a specific extension.

I am using an Elastix distro install on an Asterisk box.

I simply tried to use the Asterisk CLI and tried “dial 215”. That is not something the Asterisk CLI understands.

What can I simply do to get the Asterisk box to somehow cause that Zoiper phone to activate?

Currently the Zoiper connects to the Asterisk box which connects to the outside world via Flowroute. I have no inbound calls.

Can FreePBX do anything for me?

Why is this Asterisk stuff so obtuse!!!

What possible value is there in initiating a call from the console? I have never seen any soft switch with that capability. Where would you anchor the media stream to complete the call?

You can also get a DID from Flowroute and dial that from your wireless or landline to accomplish what you want.

Not sure that calling the platform “obtuse” is productive.

You said, “What possible value is there in initiating a call from the console?”

Ah, because it solves my immediate need (making Zoiper on the client station start an URL I need to test). Ah, because then I do not need to get a DID from Flowroute. Ah, because I do not need to get a wireless or landline involved.

When you do not know the answer, why do you instead attack people? My opinion is that Asterisk is “obtuse”. You can have whatever opinion you like about Asterisk, you do not need my permission.

Not to be further obtuse, but if you, ah, don’t need all that you don’t need, then you don’t need asterisk nor FreePBX either, you just need another sip client on another machine or ipaddress.

I am not attacking you personally. I am commenting that the idea does not seem worthy of support and it is not Asterisk that is obtuse (not even a good use of the work) but indeed the attempt to use it for an application that I still have not completely gotten my arms around.

If the need is to launch a web page on a remote client based on a SIP message I think we can find a lighter solution that Zoiper.

Many of us here of incredible depth of application experience. It is to your benefit to tell us what you are trying to do. If Asterisk/FreePBX has a solution and makes sense we can point you in the right direction. Likewise for other tools.

Right now I can say that using a bloated softphone and an IP softswitch to launch of browser is not the way to go.

If you are trying to see if telephony events can be used to trigger browser actions the answer is a resounding yes.

Zopier is an integral part of the 140+ program project. There is no way Zoiper will not be part of the solution.
Simply, I need to test and verify what the Zoiper tool claims to be able to do. That is to open an URL when Zoiper has been activated from an outside source.

There are probably 50 ways I could do this and perhaps you know as many as I do. However, they are not the issue.

I forgot to also say. . .

And, Asterisk is already a part of the solution.

Well, again I’m pretty sure that no one but you knows what the “140+ program project” is.

hmm . . . “when Zoiper has been activated” means what? it will surely be activated when you start it up, do you mean when it is sent a SIP invite, or when it is asked to register, or when it tries to register?

Decoding your definitely obtuse posts , possibly you mean the first, but as Scott suggested using an IP Softswitch to do what it was not designed to do is probably not your best choice. SIPp or sipsak or another sip endpoint (with a different ip/port than the Zoiper) will all work admirably if that is what you you are trying to do, SI (Sessions Initiation Protocol) is very well documented.

There that’s three more options for you, have you tried any one of the other 49 methods you claim to know about?

Does not help

It helps me keep smiling though :wink:

From Wikipedia

“Linguistic semantics is the study of meaning that is used for understanding human expression through language. Other forms of semantics include the semantics of programming languages . . .”

I tried the first bit on you, apparently to little effect, we still have no idea of what you are trying to build.

You will find that using the second bit properly will often clarify your meaning to others you are trying to communicate with.

  1. configuration has 1 PC as a client, 1 PC as a server and a network, if you need more details just ask
  2. client PC has a copy of Zoiper configured and running.
  3. Zoiper is configured so that upon receiving an inbound call it will open an URL
  4. Server PC is operating as an Asterisk server. It has many ways to operate. While its basic job is to connect two telephones, it can also connect, for example, the playing of a .wav file to one telephone. Another application example is custom software written to take over one leg of a conversation so that a canned operator message can be recorded on a voicemail while the operator works on a second call.
  5. I simply was asking if anyone had a simple way to get Asterisk to send electrons over the network from the Asterisk server it a proper configuration so that the already running Zoiper software believed it was receiving an inbound call.
  6. No, is a perfectly fine answer. Yes, is an even better answer.

A couple of times you have been asked for details, As yet you have a described a voip network of one extension, it has and will continually be pointed out to you that what you are doing is overkill.

Anyone who has read the documentation on asterisk and freepbx knows exactly how it works. But yes, if you want to connect for example milliwatt() to a zoiper so it can trigger an URL then use a call.file (also well documented in the body of work that is asterisk) but please be aware also that a one liner from SIPp can easily replace all that stuff.

Thanks for remembering the simple “Call File” solution. I very much appreciate it. As I recall the file needs to be copied to a specific directory and it just works. Do I need more than the following line in the “Call File”?

Channel: <SIP/315>

Or, do I need one or more of Callerid, WaitTime, MaxRetries, RetryTime, etc.?

Channel: <SIP/315>

You remember wrong, you should not copy it unless touched to the future.

I believe that yesterday was the first time in your life that you heard about it, I suggest you read up on how simple it is and you will then know the semantics involved :wink:

If you create the “Call File” in the directory Asterisk will not process the “Call File” correctly. You must copy it into the directory already completed in order for the Call File to work correctly. That’s off the top of my head. If you would like I’ll send you some links to the documentation.

This is weird, I thought you were the expert. I know I am a novice.

Thanks for the offer of the links, but that won’t be necessary, if you reread what’s off the top of your head you might rediscover that you need to mv not cp that file, which is why I said “. . .you should not copy it unless touched to the future.”

I guess I am wasting my time with you if you continue to not read digest and ultimately understand my posts, your continued passive aggressiveness is also not working ion your favor, PLEASE JUST RTFM before you reply.

As a matter of semantics, copying a file into a directory is the same as creating it there, moving it not so much as it has already been created. I guess that is still clean over your head as yet. If you look into the “top of your head” you will remember that asterisk process “very aggressively” anything with a time stamp less than or equal to now() in that directory, think about it . . .