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Other Yate server issues / Problems RTP in SIP missed-service call
« on: February 06, 2020, 11:43:10 PM »
Hello.
We have little problem with yate missed-call service over sip We configured the service in regexroute.conf file as follows:
^9999998533$=wave/play//myhome/mydirectory/missed-call-answer.alaw;autoprogress=yes;
But when we start routing a call the RTP session is not establishing. We use conventional SIP.
Also if we switch protocol settings SIP-I RTP goes well, however the CDR does not show the diversion.
Best regards,
Rail Yakupov
We have little problem with yate missed-call service over sip We configured the service in regexroute.conf file as follows:
^9999998533$=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