Asterisk Call Transfer to Analog Phone

Asterisk Asterisk Support
1

Hello Asterisk Pros,

We have an IVR setup here in our office that we’ve been working on lately. We have dial plan that is working fine, we have 1 PRI line connected to our Server (FXM8aPCI card) coming from PABX, also there are 4 analog phones connected with PABX to transfer the calls to sales/support service, we dont have SIP! only analog phone.

Right now when we use Dial() method to trasnfer the calls to analog phone, like this:

Dial(DAHDI/1/106)

it would return an error called “Cause 17 - user busy”, channel 1 is the channel on which user currently is on and listening to IVR menu, how do we just transfer the line? when we use Transfer(DAHDI/1/106) it doesn’t work!

to check this error, we plugged in another line in another channel #2, coming from pabx aswell, now when we did this:

Dial(DAHDI/3/106) it works, but then again it takes up the channel, as if another user calls and press 0 to go to extension 106, it says Cause 17 user busy.

One last issue is when the call is picked up from extension #106 we only listen a loud noise like “someone is blowing an air in the receiver!”.

Need Urgent Help!

Regards

2

Dial on a free line, typically using g.

(Asterisk doesn’t support true transfers on analogue lines. If it did, you would have to use Transfer(), not Dial(). Some people send “flash” to the line and then send DTMF digits.)

3

Can you please explain more on “dial on a free line, typically g”? Also… is the approach right? plugging an extra line from pbx to the server just to dial an extension? also how can i do call waiting/hold or parking?

4

If the call is on line 1, dial on line 2.

g searches for a free line in group of that number defined in chan_dahdi.conf.

Your final question makes me think you are completely out of your depth. I’d suggest paying for a course. If you have a purely analogue system, every leg of the call will require a line. E.g. if you have a line come in from the PSTN and go to an extension and the extension wants to transfer the call to another PSTN line, you will need two FXO lines to the PSTN and one FXS line to the extension.