Hi,

So far I have used my OpenWRT boxes only as routers and IoT gateways.
I am new to telephony, but I need to get my head around it now.

The problem:
This oh so progressive country has decided that analog phone lines are legacy technology and will be taken down.
Unfortunately the future has not quite arrived yet... So I am about to lose my phone line, but there is no fiber line to my house (and there probably never will be, unless someone digs a 15km trench just for me).
All there is is UMTS.

The solution:
Would the following work?
- UMTS dongle on a router with OpenWRT + Asterisk/chan dongle or FreeSwitch + GSMOpen to convert UMTS voice calls to/from VOIP
- A Lantuq based OpenWRT router with https://github.com/olivluca/danube-voip to convert VOIP to analog FXS
- hook up the existing analog base station to the FXS port of the router, continue to use the old DECT devices in the house.

Am I on the right track?
Is this feasible?
Or will I lose too much of the audio quality?

Usually a separate ATA (Analog Telephone Adapter) box is used for VOIP calling.  This box has an Ethernet input which issues ordinary IP packets and one or more analog phone outputs.  The CPU in the ATA box handles the VOIP protocols usually with a proprietary firmware.

Analog hardware that can be driven directly by OpenWrt is not common.

on LEDE it is possible

On all lantiq xway and xrx200 router with SLIC-120 chipsets for the FXS ports is telephone support general possible.
(other lantiq FXS chips like SLIC-121 i think too).

I know it is possible to use the FXS-ports on a Arcor EasyBox 803A Astoria ARV752DPW22
see:
https://forum.lede-project.org/t/use-um … calls/5158

(Last edited by Plonk34 on 26 Aug 2017, 22:40)

Hi Plonk34,

Thank you for your answer.
In the LEDE forum thread you linked to you wrote that you have done this already:
"The Lantiq xrx200 device O2-Box 6431 with an UMTS-Stick as an POTS-VoIP-Mobilphone-gateway is possible too."

This would be exactly what I need!

How was sound quality, when you did this? You wrote about echos - have you been able to fix them?

Is it wise to run Asterisk or FreeSwitch with the UMTS->VOIP conversion and the VOIP->LXS conversion all on the same box? Or would it be better to split these tasks across two boxes? I think UMTS to VOIP is digital to digital, and VOIP->FXS is hardware driven, so this cannot be too hungry with respect to computing power?
And an Easybox 904xDSL would even have more horsepower than the ARV752DPW22....

Like a mobilephone, no HD but clear enough.
It is more depend on the Telefon, or better the combination of fork-circuit-values and hear- and speaking-capsules.
Modern FeTAp from the 80's have no problem older have a smaller sidetone (Rückhördämpfung).
But it is possible to supress the echo:
/etc/asterisk/lantiq.conf

...
echocancel = nfees
echocancelnlp = on
...

The result was satisfactory

and you can play with the volume:

...
; The value is given in dB within the range (-24dB to +12dB), in 1 dB steps.
rxgain = 1
txgain = 3
...

Never tested but possible that will help too, is the AGC (=Auto-Gain-Control) command from Asterisk.

Yes it is wise to use it on the same box, for this reason one of the two VPE (= Virtual Processing Elements) and a littlebit RAM will reserve for the Voice process VMMC only.
If you want to separate processes it more wise to use one device as an ATA or better Telephon Central, and an other device like a pure Router with SMP support.
It is not possible to use SMP and VMMC (required for FXS) together, but it is only a problem on higher VDSL speeds < 50Mbit/s.

The delay will not better if sending the talk from station to station.

And more power consumtion.
The O2-Box 6431 have only one USB-port.
The EB-803A have an older processor but i think the only difference are the 333MHz instead 500MHz, because the software for the xrx200 devices
will compile with same parameter like the xway.