GRYNX

13th 2005f October, 2005

VoIP dect OSS Linux

by @ 15:33. Filed under Uncategorized

Important things

Contact me if you have any questions (email address at the end of this page).

Do not choose the very cheapest sound card brand, or better, check if the sound card

Stay away from the RF/high frequency/radio parts of your DECT base station. Changes there may affect your RF signal quality, detune your DECT to non-allowed frequencies, decrease range, whatever.

For non-DECT analogue phones (compared to DECT they’re really impractical, but maybe you’re staying with them for the one (privacy) or the other (EM mind control waves) tinfoil-hattish reasons ;-P), you have to make sure that the ringing circuit gets its voltage from elsewhere, so use a phone that is power by a separate wall brick and not the phone line itself. As said above, the ring-voltage-generation is not as simple as it seems.

After all, remember that I take no responsibility for these modifications!
Keine Gewähr, keine Haftung! All trademarks are the property of their respective owners.

Oh I hate disclaimers!

Links and Acknowledgements

As said above, I can only recommend this page, it explains phone <-> sound card connections in detail, as well as the required levels for phones, the most important thing when further modifications to the shown circuit ideas are needed.

Then there is this Grynx page which explains nearly all of the hardware modifications I describe here in a lot more detail.

The FLOSS product everything here depends on, Asterisk from Digium.



The phone


Front


Side


Side


Inside

Description to the notes on the last picture:

* [1]: One of the capacitors used for DC decoupling. There are two of them, but the other one is hidden below the PCB (on the right side).
* [2]: (Optional) potentiometer to reduce levels into soundcard
* [3]: The integrated circuit in this phone that does all the DECT processing and AD/DA conversion.
* [4]/[5]: Output from and input into phone.
* [6]: Connection to ring detection. There is still one transistor between the wire and [3].
* [7]/[8]: Radio front end and antenna. Do not touch this.

The phone used is a T-Sinus 2110. Some interesting thing to note about this phone:

* Cheap and good
* At least limited interoperability: Although AFAIK not advertised as GAP-compatible, it is able to connect to other base stations by pressing P followed by another (find out yourself :) button…
* It is also possible to connect other phones to the base station: Disconnect the base station from the power supply, press and hold the button on the base and reconnect power. It should be possible to search and connect with another phone now (base station PIN: 0000, if not changed). The base station will not beep as it does not contain a buzzer.

All trademarks are the property of their respective owners.

Thanks to JHH for minor corrections.


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This article has been reposted with the permission of the author.
The original article can be found here


onno@gmx.net

3 Responses to “VoIP dect OSS Linux”

  1. Serj Says:

    This is a great guide/text. Great for learning, everything covered and explained, thank you very much.

  2. phantomload Says:

    just a question, as I am not as framiliar with asterisk as you obviously are (and the documentation frankly is not upto open-source standards). But It seems it would be easier to use a voice modem with the addition of a ringer circuit (and less likely to have echo issues). As I understand this is what the x100p basically is. From what I understand an FXS is just an FXO that can make a phone ring? seems a simple addition, and there are lots of circuits available online. I do not know if you can force asterisk to route incoming voip calls to a modified FXO. Is it feasable to have astrisk actiate a parrallel port pin when the phone is supposed to ring. If this is far more difficult than what you have done. I just assumed the modem would be better matched to the phone lines than anything I build.

  3. kashif Says:

    many thx for all the efforts. I want to feed pc spkr output to pstn-analog mic input and similarly couple pc mic input with pstn spkr output. the purpose is to forward voip conversation to another pstn subscriber using pstn network.
    can anybody provide similarly simple plan.

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