Author Topic: How to configure analog phone in LinuxMCE  (Read 1408 times)

nite_man

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How to configure analog phone in LinuxMCE
« on: August 23, 2007, 12:49:49 pm »
Hi,

I have a couple Cisco 7970 phones in my installation. They works fine. Now I'm going to add doorphone which is actually an ordinary analog phone. I found a template which look appropriate for me - Generic Phone. I added the new phone using that template. But I have no idea how to configure it. The phone has its own extension but I cannot see it in Asterisk. Should I manually add that phone via AMP?
Michael Stepanov,
My setup: http://wiki.linuxmce.org/index.php/User:Nite_man#New_setup
Russian LinuxMCE community: http://linuxmce.ru

caiman

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Re: How to configure analog phone in LinuxMCE
« Reply #1 on: August 23, 2007, 04:04:26 pm »
Hi Michael,

it depends on the ATA (analog to IP) adapter that you are using. There are several options:
- if you use a sound card in your asterisk, there is a channel in asterisk based on the linux ALSA framework (I think it's called console). It should be added to AMP using the "custom" extension template.
- if you plan to use a PSTN to SIP gateway, like a sipura 1000 or higher, then all you need is to configure a generic SIP phone. You can add it in LMCE or directly in AMP as a SIP extension. A good resource to know what parameters are specific to your ATA or SIP phone is the site http://www.voip-info.org
- there may be other setups, but I haven't tried. How is your doorbell connected ?

Sam

nite_man

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Re: How to configure analog phone in LinuxMCE
« Reply #2 on: August 23, 2007, 05:06:12 pm »
Hi Sam,

Thanks for replay. The doorphone connected to the Cisco switch via FXS port. I don't have an experience with analog telephony. I configured only IP phones :)
Michael Stepanov,
My setup: http://wiki.linuxmce.org/index.php/User:Nite_man#New_setup
Russian LinuxMCE community: http://linuxmce.ru

caiman

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Re: How to configure analog phone in LinuxMCE
« Reply #3 on: August 23, 2007, 05:33:52 pm »
Hi Michael,

so I assume you have a Digium card, or similar. In that case, they are handled by asterisk through the zaptel driver, and you should be able to configure this from AMP by adding a ZAP extension.

In case you need to troubleshoot in asterisk, you can get more info here.
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels

In my case the FXS card was automatically detected and pluto had added it as a ZAP trunk. I have not used it in pluto as I moved completely to Voip, but I have been using it successfully for many years on a pure asterisk setup.

Hope this helps,
Sam

nite_man

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Re: How to configure analog phone in LinuxMCE
« Reply #4 on: August 23, 2007, 09:59:04 pm »
It's a bit different. I found some conversation about similar case - http://tinyurl.com/2p9ulv. So, my coworker who knows Cisco very well tries to configure our router similar manner.

I'm very appreciated for your help, Sam. The voip-info.org is very useful! Thanks.
Michael Stepanov,
My setup: http://wiki.linuxmce.org/index.php/User:Nite_man#New_setup
Russian LinuxMCE community: http://linuxmce.ru