Hello.
We have little problem with yate missed-call service over sip We configured the service in regexroute.conf file as follows:
^9999998533$=wave/play//myhome/mydirectory/missed-call-answer.alaw;autoprogress=yes;
But when we start routing a call the RTP session is not establishing. We use conventional SIP.
Also if we switch protocol settings SIP-I RTP goes well, however the CDR does not show the diversion.
Best regards,
Rail Yakupov
