[HELP] UK based Xorcom Rapid / TDM400P Setup

Asterisk Asterisk Support
1

Actually couple of questions… but first… here’s what I’m running:

Asterisk 1.0.9-BRIstuffed-0.2.0-RC8h (i.e. Xorcom Rapid on Debian)
TDM400P with two FXO and FXS
A Snom 360 (ext 501)
A Wifi SIP phone (can’t remember the make but it works fine, ext 502)
A normal DECT base station on one of the extensions (ext 401)
and a two other phones on the other extension (star wired to the port (ext 402)

I discovered a good discussion about getting the TDM400P working in the UK here which has helped somewhat with Q4 below.

Question 1 (ring multiple extensions on inbound call)
What I would like to do is have the Snom 360 ring for calls inbound on Zap/2 and the other phones (Ext 502, 401 and 402) ring for calls inbound on Zap/1 but after a day of total confusion trying to understand the dialplan and ‘s’ I’m stuck. Can anyone point me to the right part of the documentation or give me an example?

Looks like the thread here goes someway to explain how this can be done

The quick solution has been to add all my extensions to the first PHONE => entry in /etc/asterisk/extensions-defs.conf

and then in Rapid-menu select that entry. Every thing rings which helps make me less paranoid about missing a call.

I’m hoping I will be able to use the GotoIF function and use the ${CHANNEL} variable to detect which line the in-bound call is coming in on and then right the right extensions. See futher down this thread.

Question 2 (*# command list)
Does anyone have a list of the *# commands? I’m starting to realise that these might have to be programmed up as extensions of somekind?

Question 3 (accessing voicemail)
If I dial VoiceMailMain on ext. 201, enter the ext, #, and password, #, or if I dial VoiceMail on ext. 202, and enter the password, #, I always get the ‘Login Incorrect’ prompt. It’s driving me MENTAL. As Xorcom Rapid voicemail.conf file has:

and in \voicemail-phones.d\501.conf I have

So the VMBOX is 501, it’s password is 501, and it’s ‘full name’ is SIP Extension 501… why when I enter the correct details does it always fail?

and the answer was… because it’s not hearing the DTMF from your SIP phone. My SNOM was using inband dtmf signalling and changing that to rfc2833 (see later in this thread) it all springs to life!

Question 4 (Making it sound like it’s a UK phone system)
The answer to this is to make sure that /etc/zaptel.conf has:

and /etc/asterisk/indications.conf has:

set as it’s default location.

Question 5 (Dialing out is just plain weird)
On the Snom if I hit 9 a ‘line’ key lights up but I don’t get any dial tone. Then if I dial a number and press the ‘tick’ (i.e. dial) button I get dial tone from the BT exchange and have to re-type the number I want. What’s going on there?

If I just hit a ‘line’ button (it lights up) and I go through the same process above.

Just tested the 401/402 (zap) extensions and kinda the same thing. Pick them up, get dial tone, press nine (no dial tone), put in the number… long long pause… then get dial from the exchange… dial number… connects all ok.

Obviously I don’t want two sets of dial tone and it would be nice if Asterisk passed the numbers (silently) to the BT exchange when I get the call button. Any tips?

Question 6 (I want to be able to pick which outgoing line I use)
At the moment Asterisk seems to pick which-ever of my zap lines is available (when dialing 9 or selecting an outgoing line on the Snom) which is fine but I would like to be able to manually decide which line I want to dial out on. i.e. 9[and the number] goes out on Zap/3 and 8[and the number] goes out on Zap/4

Many thanks inadvance for any help (this is day two for me) on any of the questions even if it’s just to say ‘read such and such forum or url’.

Cheers,

sk

2

Oh boy,

All of the questions you raised are fairly easy to accomplish. But you need to do some reading. Go buy the book ‘Asterisk, the future of telephony’

To have multple extensions ring, use this

exten => 1,2,Dial(SIP/210&SIP/211&.etc…,15,omt)

The is no need to dial 9 for an outside line

You need to load the UK versions of the zaptel drivers for your TDM card(module) to get UK sounds (it’s a config exercise)

You need to program an extension to access voicemail, but try *97

Go have a look at the forums in Voipuser.org, most of the posts are UK specific and most of your questions can be answered there, but do a search first before you post.

Finally, welcome to the wonderful world of asterisk, where the learning curve is mighty steep, but rewarding.

3

Thanks for the help. I really appreciate it, especially as I know I’m being a bit cheeky posting all these questions on day two (err… three)!

I managed to get ‘the book’ yesterday afternoon. It seems pretty good (from the 30 minutes I’ve had so far). It would have been nice to have some ‘workflow’ type diagrams showing how calls come into the system and are then processed (by the example dialplan(s)).

Anyway as I get the answers I will update the thread so any other n00b users have them too!

How do I apply that to a specific Zap line only? Do I set up a context specifically for each trunk line and put a line like that in each one for the extensions I want to ring?

Anyway, looks like I’ve got plenty of reading to do!
[/quote]

4

[quote=“seskin”]Do I set up a context specifically for each trunk line and put a line like that in each one for the extensions I want to ring?

[/quote]

That’s the way i’d do it. You can also do it with dialplan logic (ie IF the call is coming in on zap/whatever, THEN do xxx) but that’s more complicated. Remember- include is your friend! put the various functions of your system (extensions, outgoing, utilities(ie voicemail), etc) in different contexts. Then make contexts for your extensions and lines and include the others as needed, lets you easily turn things on and off. Also remember- you can double include. If you include a context, any contexts included in it will also be included. IE:

[context1]
include => context2

[context2]
include => context3

[context3]
useful stuff here

if I include context1, I have also included contexts 2 and 3.

Hope that helps

5

If I use the extensions.conf can I do something like:

I’m especially interested how I tell which Zap line the call comes in on. Above I used {CHANNEL} but I’m not sure if that’s going to work? I’m guessing where I’ve put >HOMETRUNK< above it should really be an integer value. Looking in /etc/asterisk/zapata-channels.conf I have a couple of entries:

etc.

Does that channel value of 3 what I want in my entry above? Any comments?

Oh, thanks to you guys I’ve got all the extensions ringing at once now I just need to get selective about it!

I also found another potential solution on source forge AMP & Inbound routing (a solution) It suggests creating a unique context in the zap set-up for each trunk and then using that context in a Goto statement. kinda the same thing as I’m planning with the Channel statement? No?[/url]

6

Well… OK… but…

From any of the lines (SIP or Zap), if I just dial the full number asterisk appears to think I’m calling an internal extension and because there isn’t an extension at 0800-insert-some-travel-agent-number-here it gives me the engaged signal.

If I dial 9 from one of the Zap lines I get the double dial tone thingy (see above) which is just annoying but at least I can make calls.

With the Snom 360 when I dial 9 I don’t get any dial tone at all until I’ve completed the number and pressed the dial (tick) button. The Snom then displays ‘Calling’ for a moment, gives me dial tone and the word ‘Connected’… at which point it ignores any futher key presses (well… it makes DTMF noises but doesn’t appear to pass them the the BT exchange).

Thanks although I’m rather wishing I didn’t need phones at all at the moment! :unamused:

[quote="seskin"](well.. it makes DTMF noises but doesn't appear to pass them the the BT exchange).[/quote] This has been resolved by changing my sip extensions (501.conf), (502.conf) in /etc/asterisk/sip-phones.d configuration so that instead of using

it now uses

This also sorted out all my problems with VoiceMail!