Provisioning: Voice promptsAG 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"; route(__SPECIAL_DESTINATION); exit; }
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):
/usr/share/msp-media-services/prompts/msp-prompt-413.wav
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:
msp-reconfigure
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
Continue by:
- learning about Main Entities used in Provisioning; or
- learning about Provisioning: DNS; or
- learning about Provisioning: Emergency Numbers; or
- learning about Provisioning: Voice prompts; or
- learning about Provisioning: ENUM; or
- learning about Provisioning: PSTN; or
- learning about Provisioning: SIP; or
- learning about Provisioning: Trusted Peers; or
- learning about Provisioning: Voicemail; or
- going back to Provisioning Introduction.