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Diviner AG Projects RTC Platforms Provisioning: Voice prompts

Provisioning: Voice prompts
AG Projects RTC Platforms (Provisioning Guides)

This guide describes how to play voice prompts when calling specific extensions.

OpenSIPS configuration

To add a new voice prompt, edit /etc/opensips/config/siteconfig/handle-local-extensions.m4 on all SIP Proxy machines and add (example 413):

if (!DIVERTED_CALL && is_from_local() && $rU == "413") {
    $ru = "sip:800413@MEDIA_SERVER";

Additionally, should the voice prompt be always reachable, regardless of the available subscriber credit, the extension must be added to the non-blockable numbers section.

Asterisk configuration

Copy the new voice prompt in uncompressed WAV format sampled at 16KHz with mono channel to this location (msp-prompt-413.wav in this example):


Edit the configuration file /etc/asterisk/config/siteconfig/extensions-extra-prompts.m4

;  Document msp-prompt-413 purpose here...
exten => 800413,Gosub(play-prompt,s,1(msp-prompt-413,21))

Then re-generate the configuration using:


and restart the Asterisk server.

Aditional routing

To re-route calls from publicly reachable phone numbers that are routed to the platform to a voice-prompt extension, ENUM mappings can be used.

Next Steps

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