Hi, I have seen the following scenario:
Asterisk gets a call from PSTN (via Sangoma PRI card). It sends out an INVITE to the appropriate SIP peer. The SIP peer answers with 180 without SDP, but still sends some RTP packets (silent). I don't know why it does so, but it does. Asterisk now forwards the RTP packets to PSTN, thus does not generate a Ringing itself. So for the caller the line is silent until the callee picks up. According to the developers of the SIP PBX, Asterisk should ignore the RTP packets, because there is no early media session established. To me, this sounds correct. Is this a bug in Asterisk? Or is the behavior desired as it is? If not, should I raise a bug in bugtracker? We are using Asterisk 1.4.13 (because there was an issue with querying the channels via manager on Asterisks with PRI card under high load with 1.4.15), haven't tested any newer version. Best regards, Sebastian
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