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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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UK Asterisk Details

Using Asterisk in the UK


This guide describes how to get Asterisk working properly in the UK. As each telco's standards differ, services such as caller ID work differently.

Compiling Asterisk


Due to UK caller ID specifications, you will need to make a small modification to the source.
Open up chan_zap.c, and find the line:
#DEFAULT_CIDRINGS 1

Change this line to read
#DEFAULT_CIDRINGS 2

and save the file. Now build asterisk as usual.

On asterisk > 1.2 you can avoid this by adding:
sendcalleridafter = 2

to zapata.conf

BT PRI


BT is the main carrier in the UK. Their PRI lines are, by default, ISDN30e (euroisdn). By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!).

In my configuration, the PRI is plugged in to span 1 on a TE410P interface. On span 2 i have a Rhino T1 ChannelBank, and spans 3 and 4 are currently unused. My zaptel.conf is below:
span=1,1,1,ccs,hdb3,crc4
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs

bchan=1-15
dchan=16
bchan=17-31

fxols=32-55

loadzone=uk
defaultzone=uk


Even though only 8 channels are active on the PRI, all channels should be configured in zaptel.conf as shown.

The zapata.conf for the above system is shown below:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=yes
restrictcid=no
usecallingpres=no
threewaycalling=yes
callreturn=yes
transfer=yes
cancallforward=yes
echocancelwhenbridged=yes
echocancel=yes
musiconhold=default
rxgain=0.0
txgain=0.0
signalling=pri_cpe
switchtype=euroisdn
immediate=no
overlapdial=yes
pridialplan=unknown
prilocaldialplan=unknown

group=1
context = incomingfrompstn
callerid=asreceived
channel => 1-8

group=2
pickupgroup=2
callgroup=2
signalling=fxo_ls
context=dialphone
channel => 32-55


Sending Caller ID to the BT PRI


When you placed your order, you may have got a DDI allocation. If this is the case, then you may want the outgoin caller ID set to the DDI of the user making the call. You will first need to contact BT and ask them to add all your DDI numbers to the list of allowed calling id numbers. Once this is done (it takes a few hours) then you can set your outgoing callerid in extensions.conf like so:

exten => _X.,1,SetCIDNum(123456)


By default, BT pass 6 digit extensions in and out of their network.

Incoming Caller ID


Again: By default, BT pass 6 digit extensions in and out of their network.

Therefore in your incoming context, incomingfrompstn in the example here, you will need to specify these 6 digit extension.
For ease of use, i have created a macro to handle dialling a channel and passing off to voicemail:

[macro-dialddi]
exten => s,1,SetCIDNum(90${CALLERIDNUM})
exten => s,2,Dial(${ARG1},20,tT)
exten => s,3,Voicemail(u${ARG2})
exten => s,4,Hangup
exten => s,103,Voicemail(b${ARG2})
exten => s,104,Hangup



To execute this for your DDI numbers, an example incomingfrompstn context is shown below:


[incomingfrompstn]
exten => 123451,1,Macro(dialddi,Zap/32,6100)
exten => 123452,1,Macro(dialddi,Zap/33,6101)
exten => 123453,1,Macro(dialddi,Zap/34,6102)


TDM 400 FXS & BT POTS lines

(SimonHobson, 13 June 2006, mod Dirk Koopman 21 Sept 2006)

Set the card to UK lines spec by adding
options wctdm opermode=UK

to /etc/modprobe.conf (or possibly /etc/modprobe.d/zaptel or /etc/modules.conf depending on distro ?)


BT do not supply Caller ID by default, you can pay extra for this as one of the "Star Services" that are available, or, you can get it free by signing up for "BT Privacy" http://www.bt.com/privacy which signs you up to the Telephone Preference Service and gives you Caller ID for free. To set up for UK Caller ID signalling use (note: I haven't been able to verify this, it came from another forum) :
cidsignalling=v23
cidstart=polarity


In UK CID is sent after a polarity reversal and before the first ring. In theory one can set

sendcalleridafter = 0


to zapata.conf. It works for me.

FXO phantom ringing


I had a BIG problem with phantom ringing, after calls ended, there would immediately be an incoming call but you would get dialtone when you answer it. This was solved by adding :
answeronpolarityswitch=no

to /etc/zapata.conf

This should not be done for FXS lines as this will prevent CID and answering!

TDM400P & Not Detecting Hangups

Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the "Disconnect Clear Time" setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me.

"Disconnect Clear Time" is BT's name for CPC.

See http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision for more info on CPC.

Created by wiz8291, Last modification by Kevin Gearing on Wed 18 of Oct, 2006 [14:37 UTC]

Comments Filter

by Matthew Arrundale on Tuesday 12 of September, 2006 [23:45:36 UTC]
I would like to query:

"By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!). "

I'd like to know which feature needs to be turned on?

by Matthew Arrundale on Tuesday 12 of September, 2006 [23:45:30 UTC]
I would like to query:

"By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!). "

I'd like to know which feature needs to be turned on?

by Matthew Arrundale on Tuesday 12 of September, 2006 [23:45:18 UTC]
I would like to query:

"By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!). "

I'd like to know which feature needs to be turned on?

by Matthew Arrundale on Tuesday 12 of September, 2006 [23:45:09 UTC]
I would like to query:

"By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!). "

I'd like to know which feature needs to be turned on?

by Matthew Arrundale on Tuesday 12 of September, 2006 [23:44:38 UTC]
I would like to query:

"By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!). "

I'd like to know which feature needs to be turned on?

by Matthew Arrundale on Tuesday 12 of September, 2006 [23:44:22 UTC]
I would like to query:

"By default you will have no extra services turned on, so you will need to call up the ISDN management team and get them to turn on this feature (Be aware! It costs extra!). "

I'd like to know which feature needs to be turned on?

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