My ITSP provides me with a SIP trunk which requires a CallerID value for any outbound call. Though a CallerID is required, anonymous calls are allowed. See extracts from a successfull anonymous call:From: Anonymous ;tagC8b284694b5b3dePrivacy: idP-Asserted-Identi..
A recent update to the glibc-headers package (2.22-17) changed the order of members in the sockaddr_storage structure which will cause an Asterisk compile failure.We shouldnt have been relying on the order and therefore patches are up on gerrit for..
Everyone, I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The Flow is:PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISKRegular call (no srtp)works fine. However when I setup SRTP the asterisk replies with 488 Not Acceptable Here..
Dear asterisk friends,Can someone tell me whether asterisk supports PAM authenticati..