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Voice
asterisk
Commits
d0d75f61
Commit
d0d75f61
authored
10 months ago
by
Asterisk Development Team
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Update for 20.9.1
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## Change Log for Release asterisk-20.9.1
### Links:
-
[
Full ChangeLog
](
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md
)
-
[
GitHub Diff
](
https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1
)
-
[
Tarball
](
https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz
)
-
[
Downloads
](
https://downloads.asterisk.org/pub/telephony/asterisk
)
### Summary:
-
Commits: 2
-
Commit Authors: 1
-
Issues Resolved: 2
-
Security Advisories Resolved: 0
### User Notes:
### Upgrade Notes:
### Commit Authors:
-
George Joseph: (2)
## Issue and Commit Detail:
### Closed Issues:
-
819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using "make samples"
-
822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -> 20.9.0
### Commits By Author:
-
#### George Joseph (2):
-
voicemail.conf.sample: Fix ':' comment typo
-
rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
### Commit List:
-
rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
voicemail.conf.sample: Fix ':' comment typo
### Commit Details:
#### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
Author: George Joseph
Date: 2024-07-25
There can be empty slots in payload_mapping_tx corresponding to
dynamic payload types that haven't been seen before so we now
check for NULL before attempting to use 'type' in the call to
ast_format_cmp.
Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
Resolves: #822
#### voicemail.conf.sample: Fix ':' comment typo
Author: George Joseph
Date: 2024-07-24
...and removed an errant trailing space.
Resolves: #819
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