Singlebox Asterisk solution for home PBX

[Vesa]

Siemens manufactured early 2000 good chip set for home Voip adapter with versatile network support. This is Siemens Lantiq Danube platform. Several internet providers implement home network gateways Based on this chip set. Typical configuration includes:
-ADSL/DSL modem
-Wlan access point
-Ethernet switch
-ATA in form of one or two SFX port

Whole system is complete Linux based and drivers for everything is included Linux kernel. Now over decade later support is still there and major telcos in Europe are dumping DSL networks. So there is plenty of these boxes on flea market with low price.

I collected one set of these boxes several years ago and finally I has time to look them. Exact model is Elisa Kotiboksi ARV7510PW. Box is supported on Openwrt Linux distribution. Openwrt has also ready made packages for Asterisk. There is long list of Lantiq devices on https://openwrt.org/toh/arcadyan/arv7510pw.

So this harware with Openwrt offer both ATA and PBX. Supports directly analog phones and SIP trunk and endpoints. Extra item is level of ATA SFX: it is telco grade! Siemens idea was to move SFX port to telco customer home. I did measurements for SFX port. It offer full 50v DC loop voltage and 60v rms ringing voltage. Can handel several parallel electro mechanical bell ringer devices with out issue. Has hardware implemented DTFM and pulse dial decoders, ulaw, alaw and G722 audio codecs and over everything separate CPU core for Linux.

Original telco software is here totally useless, but Openwrt offer full package: boot loader, kernel and full distro. Just read instructions and re-flash whole device. I use Openwrt 19-series due low memory, Elisa version has only 32Mbytes of ram.

Package set for Asterisk after Openwrt installation is this:
asterisk16 asterisk16-chan-lantiq asterisk16-chan-sip asterisk16-codec-a-mu asterisk16-codec-alaw asterisk16-codec-resample asterisk16-codec-ulaw asterisk16-res-rtp-asterisk asterisk16-pjsip asterisk16-func-devstate asterisk16-codec-g722 nano

Simple analog to analog settings:
/etc/asterisk/

Code Select Expand

[interfaces]
;
; Number of FXS ports (default: 2)
channels = 2
;
; Create contexts per channel (default: off)
per_channel_context = on
;
[general]
;
; Gain setting for the receive and transmit path.
; The value is given in dB within the range (-24dB to +12dB), in 1 dB steps.
;
rxgain = -8
txgain = -8
; Line echo cancller valid types:
;
; off LEC and echo suppressor turned off.
; nlec LEC using fixed window; no echo suppressor.
; wlec LEC using fixed and moving window; no echo suppressor.
; nees LEC using fixed window and echo suppressor.
; nfees LEC using fixed and moving window and echo suppressor.
; es Echo suppressor
;
echocancel = nees

/etc/asterisk/extensions.conf

Code Select Expand

[from-internal]

;PHONE 2
exten = 102,1,Answer()
same = n,DIAL(TAPI/1,30,t)
same = n,Hangup()

;PHONE 1
exten = 101,1,Answer()
same = n,DIAL(TAPI/2,30,t)
same = n,Hangup()

[ltq1_out]
exten => _[+1-9]!,1,Dial(local/${EXTEN}@from-internal/n)

[ltq2_out]
exten => _[+1-9]!,1,Dial(local/${EXTEN}@from-internal/n)

[lantiq1]
include => ltq1_out

[lantiq2]
include => ltq2_out
This offers two extension numbers 101 and 102. If sip is needed, then edit pjsip.conf with suitable settings and add trunk/endpoint to extensions.conf.

[5415551212]

Very interesting thanks for sharing.
I presume you'd need at least one adapter per desired pots line.
But you'll only need to run Asterisk on one of the devices.

[Vesa]

All devices needs to run Asterisk. Easiest way is to use one box as "master Asterisk" and implement dialplan there for all phones. Then use other devices as "slave" and forward all calls to "amster" on dialplan. So this device can act as full PBX or just "dumb ATA", this can bbe done on Asterisk configuration.

One device includes two POTS lines for phones. So two phone setup needs only one box as in excample.

Here is updated link to Openwrt supported devices:
https://openwrt.org/toh/views/toh_modem_supported

Devices with CPU type "Lantiq XWAY Danube" are these ATA enabled boxes.

[5415551212]

Interesting why would all of them need to run Asterisk and not just the first one?
I would think Asterisk is not needed to work as a 'dumb' ATA.. No? 

[Vesa]

Box do not run without software. There is only Elisa propriotary firmware or Openwrt+Asterisk. Elisa firmware do not support SIP or anything else, it is just crippled ADSL modem with poor wlan.

So only way to utilize ATA interface is Asterisk with Lantiq channel drive. Due this limitations ATA adapter functionality needs whole Asterisk.

[5415551212]

Box do not run without software. There is only Elisa propriotary firmware or Openwrt+Asterisk. Elisa firmware do not support SIP or anything else, it is just crippled ADSL modem with poor wlan.

So only way to utilize ATA interface is Asterisk with Lantiq channel drive. Due this limitations ATA adapter functionality needs whole Asterisk.

Ahhh that makes sense thank you.

[Vesa]

I found second software solution for Lantiq Danube, this offer only SIP-ATA functionality. This also based on Openwrt. Software can be found here:
https://github.com/olivluca/danube-voip/releases

Seems to work correctly with Openwrt 19.07.10. Only issue is that his software do not handel two phone calls in same time, so you can not call from SFX port 1 to SFX port 2 over Asterisk. But for single line use works ok.