PSTN Phone line feature help needed

Hello all,

I’ve recently setup an Asterisk server 1.4.22-3 for our office and it is working great. It is using a OpenVox TDM400 clone card which works great with 4 PSTN lines going in. I have one minor issue that I could do with some help on.

All calls in and out work fine so far, but I have a problem with a line feature. If we need to divert our office number to another phone on our old system, I would dial 9, followed by 21 number to divert to and end with #. To remove the divert I would dial 9 followed by #21#.

When I swithed over to the new system, I transfered our calls out to a mobile from the old system while I switched over, however when it came to cancelling the divert I couldn’t on the new trixbox system. I had to fire up the old pabx, cancel the divert and then turn it off.

Basically, on the SIP phone, I dial 9#21# but as soon as it sends the first # the call fails. When I try to set a divert, it dials the whole string and then nothing happens.

What do I need to set, configure to get this to go through on the PSTN line. On the old system, I would get a dial tone first and then finish the dial string. Anyway to replicate this within Trixbox?

I don’t think it matters much, but the phones are Aastra 9133i’s.

Is there a way in Trixbox, to get a dial tone first on the PSTN line and then dial the number?

Any help would be greatly appreciated.

Yes you can get a dialtone on a DAHDI / Zaptel line by doing something like this in the dialplan: -

exten => 131,1,Dial(DAHDI/g1/)That would give a dialtone on DAHDI group 1 when you dial 131. The 131 is only a test number I have used it means nothing. I have to leave it to you to work out how you tell trixbox to do this but I’m sure it’s not difficult.

As for why exactly this happens…

Great, I will try and get this working if possible. Hopefully if I can get a dial tone then dial the number needed it will accept #21# or 21 etc.

I will let you know how I get on.

Just to clarify my understanding though does Dial(DAHDI/g1/) that portion tell asterisk to dial the DAHDI trunk G1? or is it something else? I’m wondering if I need to substiture g1 fo g0?

Still reasonably new to asterisk. Thanks again.

The phone probably has # set a “send” (or “enter” or “end-of-dialed number”, all the same in the end) so that it terminates the number at this point and sends it to asterisk. You may be able to tell the phone not tu use # as send.

You could invent your own code that dous not involve #, e.g.

exten => _121X.,1,Dial(DAHDI/g1/*21*${EXTEN:3}) exten => 122,1,Dial(DAHDI/g1/#21#)Meaning dial 121number to divert & 122 to cancel.

DAHDI/g1 meas group 1, that is all the channels (lines) on DAHDI with their group set to 1. Group are in the range 0-63 and defined in chan_dahdi.conf.

You can dial a specific line with something like DAHDI/1/. Normally there would be a number after the last / but you needed a dialtone so it’s blank.

See here for more info (zap is just the old name for DAHDI).

That’s the way I understood it. Thanks for the help. Turns out however, it was the phones local dial plan causing problems. Modifiect the local dial plan and I can now divert and cancel my diverts normally. No dial tone needed really.

Thanks again.

I thought there would be a setting on the phone. Always other ways of doing things too, which can be handy.