Hi,
I am trying to connect my BT phone and BT line to my TDM400P, but what wiring conversion do I need to do convert my phone sockets to RJ45?
Cheers,
Bill
(UK)
Hey Bill,
All you need is a normal BT-RJ11 cable, just like the ones you use for a modem. Although the RJ11 connector is smaller than an RJ45 connector, it will still fit in the socket on the back of your TDM card.
Hope this helps,
Phil.
The following link should be of use to you, this is what i used to make up a RJ45 to BT connector cable for my TDM400P FXO module.
web.ukonline.co.uk/freshwater/rj45.htm
Just a caution though, last time i checked (admittedly about a month ago) there were major issues with asterisk not being able to detect when an inbound call was remotely disconnected. Dispite digium support taking interest initially since i have provided them with all the BT documents and the British Standards documents they seem to have lost interest in this issue. This problem lead to the FXO interface being left open after a caller disconnected, and only disconnects the interface once asterisk(well zap atleast) is restarted. Maybe it has been fixed, or my configuration was bodged, never the less, just a warning.
Thanks for your replies,
I have now installed Asterisk, and got one SIP phone working with it and Asterisk ringing when there are incoming calls, but worryingly I had hangup problems like you mentioned.
Asterisk would not hang up the line when I finished a call, nor would it detect a remote hangup as you mentioned, I could hangup the line by forcing Zap to hangup using a hangup command in Asterisk. I will be having another crack at it next week.
I am new to Asterix. The main reason I was thinking of using Cisco Call Manager. However, guess what. A Cisco FXO card has similar issues. Thwere are special commands available to make it work. I have spent ages and its still not working correctly.
Its not the dialling out where the problem lies but end-call detection. There are a few ways to detect when the CALLER has hung up. One of these is to use tones and those tone frequencies must be correct for it to detect. The otehr way is to niteice that the line has dropped. But this needs additional hardware on the FXO card .Not sure they all support it.
As for Caller ID. It appears as if I can’t get that working using Cisco hardware either.
Fundementally its going to be best when everyuone uses SIP to communicate. Of course Cisco Call Manager does not support this totally at this time. …
UK is too small for people to bother right now. Things will change…otherwise you will need to switch to ISDN which I think will work fine, as this is an international standards even in the US!
jb
I remain hopeful that a solution exist, since my last message I havent really had time to have another go as yet, but I do remember from before that the caller id definitely did work with the TDM which I was quite impressed about as I didnt even expect hat aspect to work.
Hopefully in a month or so I’ll have another go, I’ll report back if I have any luck.
Surely there are enough (potential) UK users of Asterisk on POTS lines to convince the Asterisk developers that this is a needed (much wanted) problem to address?
I have the exact same problem with hangups, i have phoned digium over 10 times during the last month and they have it low on there priority list. They keep suggesting that they need multiple asterisk boxes located in the UK with remote ssh access ti troubleshoot the problem
This issue has existed since the tdm400p release and still has not been fixed. I suggest you call Chris Hozian from Digium and tell him how much UK customers need this problem resolved. If enough people report the problem it is more likely to get resolved.
This is a hurdle that is preventing my company from investing any real development into Asterisk.
Once I eventually get Asterisk setup I would be more than happy to provide ssh access to the machine for debuging uses.
Something I haven’t found out yet, does this problem plague everyone in the UK who uses analogue lines, or just some people?
I have problems with the disconnect and when reading the digium mailing lists this issue is repeated every month.
Hello,
The latest CVS HEAD of Asterisk, Libpri, and Zaptel should be able to detect remote hangup using BT and TDM400P.
You must configure the following in your zapata.conf:
- set busydetect=no
- set callprogress=yes
- set progzone=uk
Asterisk must be restarted in order for these changes to take effect.
NOTE: This will only work in the latest CVS HEAD version of Asterisk, Libpri, and Zaptel.
Regards,
Chris Hozian
Humm I was having hang up problem on my TDM400P
This is some info from my zapata.conf
usecallerid=yes
cidsignalling=v23
;cidstart=polarity
hanguponpolarityswitch=yes
;busydetect=yes
;busycount=6
and also I found that putting
modprobe wctdm opermode=UK in my zaptel file also helped
I have callerid working and it hangs up the phone ok I am not saying this is the correct way but it’s working for me
Tim
Having read this thread, it would appeat that not a lot of people are having much success with using TDM400P and Asterisk in the UK - is this correct???
Have I wasted time and money trying to provide a telecomms solution that is not going to work at all?
Craigad
I will say that it’s not easy to find the information.
But its out there plenty of googling ect.
My final config that is working for me and I am in the uk is
[code][channels]
language=en
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
usecallerid=yes
ukcallerid=yes
;cidsignalling=v23
;cidstart=polarity
;hanguponpolarityswitch=yes
busydetect=yes
;busycount=6
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=3.0
txgain=3.0
group=0
callgroup=1
pickupgroup=1
immediate=no[/code]
now nothing to say this is the best settings but it is working and I am getting caller ID.
Also take a look http://www.pbxforum.co.uk/forum/showthread.php?t=68
drbig,
thankyou very much for the info! You have re-assured me that I have made the right choice for our small company.
Much appreciated.
Just as a matter of interest, do you use Asterisk@Home or have you installed Asterisk using cvs or tarballs direct from asterisk?
/Craigad
Have done both !!!
Started with a ground up build then had a play with asterisk@home.
Use a@H for test machine as can blank and install quickly
ground up build for production as a@h has some things loaded that I dont want or need.
all mods or changes are tested on the a@h machine before putting on the live server.
Tim
Now that makes a lot of sense!
Thankyou very much for your help with this.
I’ll keep you posted on my progress.
/Craigad
As promised - a follow up.
I have now got Asterisk working on both inbound and outbound calls through my TDM400P.
Had a few problems with outbound routing - I had asterisk use Zap/g1 and I was getting connected but silence.
Change the Zap trunk ID to Zap/g0 and everything works wonderfully now.
Thanks for your help
/Craigad
hi carigad,
I am also implementing an asterisk server in the UK, though doing all of it remotely. anyway my system uses a sangoma a200 card and one BT line. All have been configured and setup, however, when I place an outgoing call, the CLI shows that it gets connected but only silence (like what you have describe in your scenario) can you provide some inputs or advice on how were you able to surpass the dilemma? Also, would it be ok if you post your configs (zapata, zaptel, extension)? I would just like to compare it with mine if ever I missed out on things.
Many thanks.