Forum Replies Created
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> Isn’t calling 56781 from another extension an inbound call?
Nope, it is standard outbound call.
If you want to test inbound (incoming) call flow you should call the PBX from an outside line (trunk), or, you can trick the PBX by changing the Context the Extension will execute when you make the call.
By default all extensions will execute from-internal context but you can can change it to from-pstn and Elastix will treat all calls from that extensions like outside calls (inbound calls).
Try it with one extension to get an idea.Best, Martin
OK, so maybe I don’t understand call flow like I thought I did. Why is it I can put the wait statement in the outbound_vdp and get the desired result. Isn’t calling 56781 from another extension an inbound call?
I’m using a different computer and the clipping issue seems to be resolved. However I still cannot implement a wait/pause before the playback occurs.
Strange, I am using the same Elastix version with VDP and all works just fine, no sound files clipping…
I guess you should look at your Elastix configuration.
Just my 2c.Best, Martin
No problem. Enjoy!
Kevin
For the moment i use the context named ‘from-sip-external’ to write my dialplan until i figure out what to change in the configuration.
Anyway, thanks for your help.
Kyriakos
That’s correct. If Asterisk doesn’t know WHERE you want to send an inbound call (the context. And there can be many of them or just 1 to handle all calls) then it won’t know what to do with the call and it will fail and usually give a “number not in service” type message.
Kevin
Ok this is what i understand so far:
There is no problem with me having compiled Asterisk for the dongle to work.
There is no problem with me having no inbound route and no Voip trunkBUT i have to make an inbound route to send incoming calls (from?) to …vdp-inbound context?
That’s because that’s the context asterisk is looking for when an external sip call arrives. You can change that in the asterisk config for inbound routes if wanted. This has more to do with your asterisk config than it does visual dialplan.
Kevin
No problem James. Enjoy VDP! It’s really a great tool!
Kevin
I changed the config in the server to persistent/etc/asterisk.conf and now it connects,
Thanks Kevin.James,
As this is an embedded device utilizing a special version of linux (uclinux) there is a very good chance that the asterisk installed on the system uses a custom, non-standard directory layout. You might try searching the device for the asterisk.conf file and point to that directory for the “Config file” setting in your server definition and see if that works. Let me know how you make out.Kevin
Thanks Mat,
You confirmed that I needed “_.” not just “_” and it know works with gsm/wav files 🙂
Hi there,
The first block in hello-world dial plan is the Extension that will handle that dial plan.
The extension number is 56781.
This mean you have to dial 56781 to run that dial plan, and to hear hello worlds sound file.
Did you try calling 56781?If you want to run that dial plan for all numbers, replace 56781 with “_.”, not only “_”.
— Mat
Bob,
There are actually 2 parts here. The license for VDP itself is tied to the MAC address of the ethernet adapter on the machine you are working on. Depending on how many licenses you purchase you may use VDP to access more than one server. The initial purchase allows you to connect to 1 server when activated. However, you are able to deactivate a license on one server and activate it on another, you just can’t access both at the same time unless you purchase another license. So you would be able to develop on a test box and when everything worked the way you want you could unregister the test box and register on the production box and then deploy. This is assuming you have the same version of Asterisk, etc.Now, the other thing is, that any code created in VDP is nothing more than standard asterisk dialplan code and can be written on one system and deployed on others just by taking the code generated and manually adding it to another server and modifying your contexts on that server to properly call your dialplan. So in theory you only need to use 1 license for VDP to create any code you want and then deploy it anywhere you want by manually integrating the generated code with your PBX. I have done this before and it works fine. You don’t HAVE to use VDP to deploy your finished product if you don’t want to. Last time I used Elastix it was basically just another distro of Asterisk with FreePBX and other components integrated in the system, so the above information should apply to Elastix as well. If I’m wrong I’m sure support will correct me. 🙂
Kevin