I have FreePBX 14 (Asterisk 13) installation, connected to PSTN over SIP with Cisco2911 with FXO cards.
I need playback message to called person, and I try do it by adding to trunk dial option A(message).
The problem is that message begins to play when Asterisk connected to Cisco, not when called person pick up the phone.
I think this is becouse Asterisk receive SIP status 200 OK when he connect to gateway, not to called person.
Is it possible to do so Asterisk understand when real human pick up the phone? Or, may bee, configure a Cisco to send SIP status 180 Ringing until PSTN phone not answered?
Most analog PSTN lines do not provide “answer supervision”. If your carrier does this by polarity reversal, your FXO cards may handle it; see https://www.cisco.com/c/en/us/products/collateral/interfaces-modules/voice-modules-interface-cards/product_data_sheet0900aecd8016c1c6.pdf .
If this is an incidental use of your PBX, the easiest option may be to use an inexpensive SIP provider for just those calls. What country are you calling? Fixed or mobile? Approximate call volume? Average message length.
Depending on your application, another possibility is to play a short message repeatedly, e.g. “Press 1 to hear your message”, detect the DTMF and then play the real message.
Also, even if you get answer supervision working properly, you may still have issues with calls going to voicemail, out-of-range announcements, etc.
For better advice, please describe your application.
What country are you calling? Fixed or mobile? Approximate call volume? Average message length.
Russia, fixed line, message 2-3 min. length. We dont have problems with outgoing call, only with play message: it starts before human pick up the phone.
The message - notification that the call can be recorded. The notification should be played only once, then the operator should start the conversation.
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