The bug described here https://issues.asterisk.org/jira/browse/ASTERISK-18345 is still not fixed in asterisk-10.1.3. It renders SIPS support virtually unusable (at least with Snom phones). Please add the attached patch to the patchset for 10.1.3. Reproducible: Always Steps to Reproduce: 1. Setup SIPS for phone 2. Try to call a number Actual Results: SIP connection is dropped Expected Results: Call should succeed
Created attachment 304907 [details, diff] Fix for the TLS problem from ASTERISK-18345 See https://issues.asterisk.org/jira/browse/ASTERISK-18345 for the upstream bug report.
+*asterisk-10.2.0 (13 Mar 2012) + + 13 Mar 2012; Tony Vroon <chainsaw@gentoo.org> +asterisk-10.2.0.ebuild: + Add correct divisor to trunk frequency for IAX2 channels, from an upstream + commit by seanbright. Chained certificate support & correct handling of + non-blocking I/O for TLS/SSL, as reported by Matthias Nagl in bugs #407781 & + #407919. Upstream has fixed the port number in outbound SIP NOTIFY packets, + included iLBC, fixed the caller ID in originated calls and stopped UDPTL from + being created unneccesarily. Also the SIP timer should no longer be stopped + prematurely.
