Hi
i have two asterisk servers connected though SIP trunk
when user X at Asterisk server one call user Y at Asterisk server two using
DIAL with m option.
Dial(SIP/${ext...@asterisk_trunk_2,30,m)
IF Y is un registered and have voicemail.
User X will not hear the voicemail prompt to enter the message 183 early
media .
How can I replace the music RTP stream sent to the client with the early
media RTP stream coming from Asterisk server2 voicemail since there is
already an early media
Generated by MOH.
NB that I don't want to start billing until user hear the beep
That can be done by editing chan_sip.c or app_dial.c ?
regards
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