The release of Asterisk 13.38.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
| pjsip: Crash on call rejection during high load | (Reported by Sandro Gauci) |
Improvements made in this release:
-----------------------------------
| Increase reg_server column size for ps_contacts table realtime | (Reported by sungtae kim) | |
| Create a Bridge with video_single mode | (Reported by sungtae kim) |
Bugs fixed in this release:
-----------------------------------
| res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies | (Reported by Sebastian Damm) | |
| resource_endpoints.c : Memory leak if endpoint not found | (Reported by Jean Aunis - Prescom) | |
| res_pjsip_config_wizard: Crash when freeing string when failing to add extension | (Reported by Vieri) | |
| app_voicemail: Undocumented behavior from VMSayName | (Reported by Eric Smith) | |
| res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used | (Reported by Sebastian Damm) | |
| dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format | (Reported by �家建) | |
| Music On Hold announcement cuts intro of music the first time it is played | (Reported by Thomas Frederiksen) | |
| res_stasis: Add compare function for bridges moh container | (Reported by Hajek Michal) | |
| func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT | (Reported by Péter Juhász) | |
| Unable to get rtp codec payload code for slin | (Reported by Brian J. Murrell) |
New Features made in this release:
-----------------------------------
| Implement support for History-Info | (Reported by Torrey Searle) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.38.0
Thank you for your continued support of Asterisk!
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