I want to set up my kamailio SIP server to answer calls and playback a message instead of replying to INVITE requestes with BUSY.
I setup an asterisk server that plays back the message to all incoming calls, then tried these things:
- set this server as
voicemail.srv_ip, enabled WITH_VOICEMAIL, setvoicemail.srv_port- but no audio stream is set up from asterisk, so the call doesn’t hear the message. I presume this is by design? (ie: kamailio assumes the voicemail only receives audio)
- set this server as a low priority dispatcher
- but it cannot handle multiple concurrent calls
Any thoughts on how to do this with kamailio and asterisk (or just kamailio if that is possible) greatly appreciated!
For a solution with Kamailio only, check the rtp_media_server module, it in alpha stage, but maybe (with a bit of hammering) it offers what you need.
Using Asterisk (or other media server) is the classic solution, it should work quite easy, pretty much every deployment out there is a combination of Kamailio with a media server. If you do not get audio, then it is not Kamailio preventing that, because Kamailio does not deal with media. Maybe you need to do NAT traversal and use an RTP relay application such as RTPEngine or RTPProxy.
As of using dispatcher, definitely there is support for concurrent calls, even I am not sure if you referred to Kamailio or Asterisk, both of them support many calls in parallel.