diff options
| author | Sebastian Kemper <sebastian_ml@gmx.net> | 2019-07-12 20:23:11 +0200 |
|---|---|---|
| committer | Sebastian Kemper <sebastian_ml@gmx.net> | 2019-07-12 20:23:13 +0200 |
| commit | 1fd2890531e0668c0c7b85847dd09f3fe4aceb94 (patch) | |
| tree | 250f5c15d77822bbbd5c8ae0afbf5398dda8e33b | |
| parent | cc0658c8533b870de1deed4a656edb8bacf23fb6 (diff) | |
asterisk-15.x: fix AST-2019-003
https://downloads.asterisk.org/pub/security/AST-2019-003.html
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
| -rw-r--r-- | net/asterisk-15.x/Makefile | 2 | ||||
| -rw-r--r-- | net/asterisk-15.x/patches/160-AST-2019-003-15.diff | 39 |
2 files changed, 40 insertions, 1 deletions
diff --git a/net/asterisk-15.x/Makefile b/net/asterisk-15.x/Makefile index 67ff71e..a73fbc9 100644 --- a/net/asterisk-15.x/Makefile +++ b/net/asterisk-15.x/Makefile @@ -9,7 +9,7 @@ include $(TOPDIR)/rules.mk PKG_NAME:=asterisk15 PKG_VERSION:=15.3.0 -PKG_RELEASE:=5 +PKG_RELEASE:=6 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases diff --git a/net/asterisk-15.x/patches/160-AST-2019-003-15.diff b/net/asterisk-15.x/patches/160-AST-2019-003-15.diff new file mode 100644 index 0000000..0c8f89a --- /dev/null +++ b/net/asterisk-15.x/patches/160-AST-2019-003-15.diff @@ -0,0 +1,39 @@ +From a8cc63a8b2b973d6d34251d74b8d4576d6796dce Mon Sep 17 00:00:00 2001 +From: Francesco Castellano <francesco.castellano@messagenet.it> +Date: Fri, 28 Jun 2019 18:15:31 +0200 +Subject: [PATCH] chan_sip: Handle invalid SDP answer to T.38 re-invite + +The chan_sip module performs a T.38 re-invite using a single media +stream of udptl, and expects the SDP answer to be the same. + +If an SDP answer is received instead that contains an additional +media stream with no joint codec a crash will occur as the code +assumes that at least one joint codec will exist in this +scenario. + +This change removes this assumption. + +ASTERISK-28465 + +Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87 +--- + +diff --git a/channels/chan_sip.c b/channels/chan_sip.c +index fe2ae1e..6251878 100644 +--- a/channels/chan_sip.c ++++ b/channels/chan_sip.c +@@ -10921,7 +10921,13 @@ + ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0)); + } + +- if (portno != -1 || vportno != -1 || tportno != -1) { ++ /* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or ++ * video is not being transported, thus we continue in this function further up if that is ++ * the case. If we receive an SDP answer containing both a UDPTL stream and another media ++ * stream however we need to check again to ensure that there is at least one joint codec ++ * instead of assuming there is one. ++ */ ++ if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) { + /* We are now ready to change the sip session and RTP structures with the offered codecs, since + they are acceptable */ + unsigned int framing; |