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...Elazar Broad) * ASTERISK-24019 - When a Music On Hold stream starts it restarts at beginning of file. (Reported by Jason Richards) * ASTERISK-24211 - testsuite: Fix the dial_LS_options test (Reported by Matt Jordan) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) Improvements made in this release: ----------------------------------- * ASTERISK-24171 - [patch] Provide a manpage for the aelparse utility (Reported by Jeremy Lain??) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/aster...

...hards) * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying if ever not able to resolve (Reported by David Herselman) * ASTERISK-24211 - testsuite: Fix the dial_LS_options test (Reported by Matt Jordan) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) * ASTERISK-23577 - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by Jay Jideliov) * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky)...

...Elazar Broad) * ASTERISK-24019 - When a Music On Hold stream starts it restarts at beginning of file. (Reported by Jason Richards) * ASTERISK-24211 - testsuite: Fix the dial_LS_options test (Reported by Matt Jordan) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) Improvements made in this release: ----------------------------------- * ASTERISK-24171 - [patch] Provide a manpage for the aelparse utility (Reported by Jeremy Lain??) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/aster...

...hards) * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying if ever not able to resolve (Reported by David Herselman) * ASTERISK-24211 - testsuite: Fix the dial_LS_options test (Reported by Matt Jordan) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) * ASTERISK-23577 - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by Jay Jideliov) * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky)...

...ated during dial operation (Reported by Matt Jordan) * ASTERISK-24231 - crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable (Reported by Niklas Larsson) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) * ASTERISK-23577 - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by Jay Jideliov) * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky)...

...ated during dial operation (Reported by Matt Jordan) * ASTERISK-24231 - crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable (Reported by Niklas Larsson) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) * ASTERISK-23577 - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by Jay Jideliov) * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky)...

..._rtp_state when RTP instance is NULL (Reported by Jay Jideliov) * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod) * ASTERISK-24181 - RLS: Large lists don't get sent because they exceed the PJSIP message length limit (Reported by Jonathan Rose) * ASTERISK-24254 - CDRs: Application/args/dialplan CEP updated during dial operation (Reported by Matt Jordan) * ASTERISK-24241 - crash: CDRs r...

.../jira/browse/ASTERISK-23634>] - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls (Reported by Roman Skvirsky) - [ASTERISK-24249 <https://issues.asterisk.org/jira/browse/ASTERISK-24249>] - SIP debugs do not stop (Reported by Avinash Mohod) - [ASTERISK-24181 <https://issues.asterisk.org/jira/browse/ASTERISK-24181>] - RLS: Large lists don't get sent because they exceed the PJSIP message length limit (Reported by Jonathan Rose) - [ASTERISK-24254 <https://issues.asterisk.org/jira/browse/ASTERISK-24254>] -...