I used following schematic



Additionally i connected three twisted pair wires with a 100 Ohm resistor. At the end of each wire a schottky diode reverse biased across the DS18S20 sensor against spikes is used.

At the moment i have 5 DS18S20 temperature sensors (inside, outside, heater back, heater, warmwater)

Initialising: digitemp -i -s /dev/ttyS0 -q -c /etc/digitemp.conf

Reading: digitemp -a -q -c /etc/digitemp.conf

Mar 16 12:44:06 Sensor 0 C: 20.94 F: 69.69
Mar 16 12:44:07 Sensor 1 C: 13.75 F: 56.75
Mar 16 12:44:09 Sensor 2 C: 28.69 F: 83.64
Mar 16 12:44:10 Sensor 3 C: 31.94 F: 89.49
Mar 16 12:44:11 Sensor 4 C: 48.75 F: 119.75

/etc/collectd.conf:
LoadPlugin exec
<Plugin exec> Exec "nobody:dialout" "/usr/bin/digitemp_DS9097" "-n0" "-d10" "-a" "-q" "-c/etc/digitemprc" "-oPUTVAL home/temp/temperature-%s interval=10 %N:%.2C"
</Plugin>

I had to make a patch for collectd that the exec plugin runs with user nobody and group dialout. Because user nobody's group is not dialout "setgroups" have to be called (add user nobody to /etc/groups dialout is only used if initgroups is called).

Also for digitemp i made a patch because i got sometimes 85 degrees (error condition) which are not correctly handled by digitemp:

Hi!

I discovered the use of wrong indexes:

These is your Indextable:

Temperature = scratchpad[1]
Sign = scratchpad[2]
TH = scratchpad[3]
TL = scratchpad[4]
Count Remain = scratchpad[7]
Count Per C = scratchpad[8]
CRC = scratchpad[9]


1. glitch condition

COUNT_PER_C is Index 8 and COUNT_REMAIN is Index 7

- if( (scratchpad[7] - scratchpad[6]) == 1 )
+ if( (scratchpad[8] - scratchpad[7]) == 1 )

2. Error Condition (85 Degrees)

- if( (scratchpad[4]==0xAA) &&
- (scratchpad[3]==0x00) &&
- (scratchpad[7]==0x0C) &&
- (scratchpad[8]==0x10)
+ if( (scratchpad[1]==0xAA) &&
+ (scratchpad[2]==0x00) &&
+ (scratchpad[7]==0x0C) &&
+ (scratchpad[8]==0x10) )

Also i think an Error Condition (85 Degrees) should be handled like a
CRC Error.


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I will make calls from my phone to PSTN and with dial extension "#9" to my voip account.

This is done with following dialplan "(xx.<:@gw0>|<#9,:>xx.)"

If a call from PSTN is not answered within 20 sec the call should be redirected via voip (see CFNA Solution).

Because my SPA-3102 is behind a firewall i use it as a bridge.

Firewall

modprobe ip_conntrack_sip ports=9060; modprobe ip_nat_sip
iptables -A FORWARD -p udp --dport 9060 -m state --state NEW,ESTABLISHED -j ACCEPT

With this settings i don't need "NAT Keep Alive" and Register every 3600 sec (1 hour) is ok.

Usually udp connectiontracking has a timeout of 180 sec, but ip_conntrack_sip reads the sip content and set the timeout (3600).

My Settings

WAN

Connection Type: DHCP
Enable WAN Web Server: yes

Lan Setup

Networking Service: Bridge
Enable DHCP Server: no

SIP

RFC 2543 Call Hold: no
RTP Port Min: 9000
RTP Port Max: 9059

Provisioning

Provision Enable: no
Upgrade Enable: no

Regional

Ring1 Cadence: 60(1/4)
Ring Waveform: Sinusoid
Ring Frequency: 50
Ring Voltage: 60
CWT Frequency: 420@-20
FXS Port Impedance: 270 + (750 || 150nF)
Caller ID Method: ETSI FSK
Caller ID FSK Standard: v.23

Line 1

Make Call Without Reg: Yes (if no network connected)
Proxy: 1.2.3.4:9060
User ID: sipuid
Password: xxxx
Preferred Codec: G729a
G7* Enable: no
Dial Plan: (xx.<:@gw0>|<#9,:>xx.)

PSTN Line

Make Call Without Reg: Yes
User ID: sipuid
Password: xxxx
Preferred Codec: G729a
G7* Enable: no
PSTN CID For VoIP CID: Yes
medium Min CPC Duration: 0.25
Disconnect Tone: 440@-30,440@-30;2(0.4/0.4/1+2)
FXO Port Impedance: 270 + (750 || 150nF)
On-Hook Speed: 3ms
Line-In-Use Voltage: 25
Current Limiting Enable: Yes
Ring Validation Time: 150 ms
Ring Indication Delay: 0
Ring Timeout: 650 ms

Without this "Ring Timings" after "PSTN Ring Timeout" the call to Line 1 got CANCELED also immediately the following call to VOIP (don't know why - saw it in traces to syslog).
With "PSTN Ring Timeout: 20" i also saw no cancel but that's a dirty workaround for me (better are a correct "Ring Timings").

Enable Traces

System

Syslog Server: 1.2.3.4
Debug Server: 1.2.3.4
Debug Level: 3

Line 1 / PSTN Line

SIP Debug Option: full

CFNA Solution 1:

User 1

Cfwd No Ans Dest: 12345678
Cfwd No Ans Delay: 20

PSTN Line

PSTN Answer Delay: 40 (20 sec Line 1 and 20 sec voip => after that forced hangup)

disable via phone with *93
enable via phone with *92[Number as dialed via dialplan(use #9 for voip)]

Drawback:
* after disabling with *93 the number is deleted
* enable with *92#9NUMBER (without #9 PSTN is used according to Dialplan (xx.<:@gw0>|<#9,:>xx.) it would be NUMBER@gw0)
* only G711 Codec supported (because PSTN calls Line 1 and then only G711 seams to be supported)

CFNA Solution 2:

PSTN Line

Dial Plan 8: (S0<:12345678@1.2.3.4:9060>)
PSTN Caller Default DP: 8
PSTN Answer Delay: 20


Drawback: you cannot dis/en-able via phone

CFNA Solution 3:

Use a local asterisk where you can connect sip clients rinnging at the same time when a call arrives on the SPA-3102

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copy local

cp

cp -dpRx /OLD /NEW

tar

cd /OLD && \
tar -cp --atime-preserve --one-file-system --exclude=/lost+found -f - . \
| ( cd /NEW && tar -xpv --atime-preserve -f - )

rsync

rsync -avHx --delete /OLD /NEW

copy via network

tar

cd /OLD && \
tar -cp --atime-preserve --one-file-system --exclude=/lost+found -f - . \
| ssh 192.168.0.1 "( mkdir -p /NEW ; cd /NEW \
&& tar -xpv --atime-preserve -f - )"

rsync

rsync -avHx --delete /OLD/ root@192.168.0.1:/NEW

rsync -avz -e ssh remoteuser@remotehost:/remote/dir /this/dir/

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mkdir /boot/dos
cd /boot/dos
cp /usr/lib/syslinux/memdisk .
wget http://www.fdos.org/bootdisks/autogen/FDOEM.144.gz
gunzip FDOEM.144.gz

mkdir /mnt/temp
mount -o loop -t vfat FDOEM.144 /mnt/temp
copy new bios-image and flasher to /mnt/temp
umount /mnt/temp

/boot/grub/menu.lst:

title DosDisk
kernel /boot/dos/memdisk.bin
initrd /boot/dos/FDOEM.144

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after upgrade from feisty to gutsy firefox and gimp wont work:

the solution is

dpkg-reconfigure libcairo2 libpango1.0-common
fc-cache -fs
update-pangox-aliases

http://www.mail-archive.com/debian-user ... 03742.html

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