Re: Dial plan Config with a twist

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Re: Dial plan Config with a twist

Postby ola007 » Mon Dec 21, 2015 7:43 am

Hello there my asterisk friends and professionals.

I am using asterisk asterisk on an armhf machine without any audio device purposefully, I am using it in conjuction with some devices that enable SIP calling but they dont have dial key pads. So the question is how can I create a dial plan that would enable me control the SIP devices from the console remotely? I am able to register the calling devices, and I have been able to call from a regular laptop to the devices without problems. I am trying to use direct media set up, so that when I dial an extension from my arm it connects both caller and callee . The diffuclty is that the SIP devices have no dial keys as stated. So I am looking to see how to initiate a call from one of my devices to another from the CLI, I tried to use the originate function to brigde my devices, but once the destination rings from the remote party, there is a hangup and am left with the PBX to the other device. I need a "device to device" directly from CLI configuration. Any help would be very much appreciated. Thank you and happy holidays!
And in one of my SIP.conf I know I register my devices as I would a phone, but do I register the devices am calling as well, I thought the server conncetivity would take care of the other device, network set is fine. Thanks!
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Re: Dial plan Config with a twist

Postby david55 » Tue Dec 22, 2015 2:05 am

You need to register any device that you want to call but for which you don't know the IP address. You don't need to register type= friend devices from which you only receive calls, and you don't need to register if you already know their IP address. Registering and using type=peer is normally more secure than relying on type=friend for dynamic devices.

I can't work out why you would get a premature hangup with your originate, so you need to provide the relevant diaplan, the originate command, and verbose level 5 logging.
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Re: Dial plan Config with a twist

Postby david55 » Tue Dec 22, 2015 2:10 am

Also, are you aware that if the device initiates a call with no user in the URI, Asterisk will use the s extension?
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