How to comunicate a PABX(Analogic) with Asterisk

Hello,

I’m interrsteing about to communicate a traditional PABX with Asterisk.
I’ve already a runnig PABX, an I want to use Asterisk too.

I was blocked when I’ve tried to configure the IVR.

is there Anybody can help me?

Best regards.
:wink: :smile:

iantra

well, for starters, you could post what type of legacy PBX you’re using, along with how you’re connecting to asterisk and all that.

then, to really help, you could post the output of the CLI when the communication fails…

[quote=“iantra”]Hello,

I’m interrsteing about to communicate a traditional PABX with Asterisk.
I’ve already a runnig PABX, an I want to use Asterisk too.

I was blocked when I’ve tried to configure the IVR.

is there Anybody can help me?

Best regards.
:wink: :smile:

iantra[/quote]

Ok, what IVR scripts fo you want?

Hello,

The PBX what I use is a Nortel M6501-R.
The phone is a AASTRA M740E (Anthracite blue).

I created the Dial Plan, then i can call the AASTRA phone with a X-Lite, Idefisk.

But the problem appear when the AASTRA phone call one Xlite Client.

The call can be passed through the PABX but when Asterisk ask to type an extension (type-ext-of-person), he can’t receive the number typed with the AASTRA phone.

The Dial plan is correct, because the call can be done with Wlite or Idefisk :confused:

I don’t know why the extension requested don’t have been received by Asterisk.

Best regards.

:smile:

I am not an expert, there are astmasters like IronHelix, baconbuttie and others (they are admirable). However, what I assume from your statement is that your asterisk box could not send the dtmf signals. Did you try by forcing the DTMF in the box? If not, add the following:

dtmfmode=auto

in the general section in your sip.conf. And do not forget to either restart the server or reload the sip conifguration by typing “sip reload” in asterisk CLI.

Hopefully it helps!

[quote=“zenny”]I am not an expert, there are astmasters like IronHelix, baconbuttie and others (they are admirable). However, what I assume from your statement is that your asterisk box could not send the dtmf signals. Did you try by forcing the DTMF in the box? If not, add the following:

dtmfmode=auto

in the general section in your sip.conf. And do not forget to either restart the server or reload the sip conifguration by typing “sip reload” in asterisk CLI.

Hopefully it helps![/quote]

Thank you, I will try this idea. :wink:
You are very helpfull