FreePBX Voicemail for Siemens Hicom 150E

Hi All.

We have FreePBX / Asterisk being used very successfully in a production environment, with connection to VoIP providers via SIP and IAX trunks, SkypeForAsterisk, IVRs and DISA with Cepstral, many extensions etc. We have also connected our existing Siemens analogue PABX to Asterisk using a PSTN FXO analogue gateway using two stage dialing via SIP extensions, so we can route calls to and from Asterisk and the Siemens PABX. All working very well. What we want to do now is replace our existing and old voicemail system which connect to the Siemens PABX with the voicemail module that comes with FreePBX.

Current system

  1. Someone external calls a DID number handled by our Siemens PABX.
  2. The call goes to a Siemens extension and rings, and no one answers.
  3. The call then gets switched to a POTs extension on the Siemens which connects to our old voice mail unit.
  4. The Siemens sends some DTMF codes to tell the voicemail unit which extension is calling for example - 14441
  5. Call goes to appropriate voice mail box in voicemail unit.

New system

I want to unplug the POTs lines that goes from the Siemens PABX to the voicemail unit, and connect them to another FXO gateway, then connect this to Asterisk via SIP Trunks and use 1 stage dialing.

I don’t know how to pick out the dtmf tones sent by the Siemens PABX and use these to tell Asterisk which extension to send the call to so we can use the FreePBX voicemail module.

It would be really good if the Siemens sent the CID from the extension receiving the call, eg: 504, so that can be used to route the call, but I’m not sure if it does, bit new to this all.

Any ideas anyone ?



Hi SkykingOH.

Thanks for replying. I don’t suppose you could point me in the right direction regarding where to put the dial plan code and what the code entails ? I’m imagining it will go in sip_custom.conf ? or one of those files. I haven’t had much to do with dial plan code, but i want to learn more. Also as far as i know the lines connecting to the existing voicemail box are PSTN or POTs lines, and i have the call termination all sorted out using the FXO gateway.



This is not a simple task and can’t be handled directly with FreePBX. You will need to write Asterisk dial plan code to pick out the DTMF digits, format them and then hand off to the voice mail module.

Also the handshake with the Siemens will be challenging if it simply comes off hook and sends the digits? Are you sure these aren’t TIE lines, if so they will expect battery from the Asterisk box and you need to use FXS interfaces.