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| author | Jiri Slachta | 2019-04-09 17:16:42 +0000 |
|---|---|---|
| committer | GitHub | 2019-04-09 17:16:42 +0000 |
| commit | 867b79bf755cb587a5c7f9628121c0b8edc83a60 (patch) | |
| tree | cd7e39f9ed7681578dd4d50b4a20c6a45b86bc56 | |
| parent | 9a8f3b372610bb9dc2bdbe58328f72125b15d200 (diff) | |
| parent | 97c00a118c492db3d5857bb3a4d8846e3796d69c (diff) | |
| download | telephony-867b79bf755cb587a5c7f9628121c0b8edc83a60.tar.gz | |
Merge pull request #414 from micmac1/AST-2019-001
asterisk-15.x: add patch for AST-2019-001
| -rw-r--r-- | net/asterisk-15.x/Makefile | 2 | ||||
| -rw-r--r-- | net/asterisk-15.x/patches/110-AST-2019-001-15.diff | 34 |
2 files changed, 35 insertions, 1 deletions
diff --git a/net/asterisk-15.x/Makefile b/net/asterisk-15.x/Makefile index 5d4ed3e..af7135a 100644 --- a/net/asterisk-15.x/Makefile +++ b/net/asterisk-15.x/Makefile @@ -9,7 +9,7 @@ include $(TOPDIR)/rules.mk PKG_NAME:=asterisk15 PKG_VERSION:=15.7.0 -PKG_RELEASE:=1 +PKG_RELEASE:=2 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases diff --git a/net/asterisk-15.x/patches/110-AST-2019-001-15.diff b/net/asterisk-15.x/patches/110-AST-2019-001-15.diff new file mode 100644 index 0000000..f7a68be --- /dev/null +++ b/net/asterisk-15.x/patches/110-AST-2019-001-15.diff @@ -0,0 +1,34 @@ +From 476d60f850c75ca9142aaf783992db74efea6a49 Mon Sep 17 00:00:00 2001 +From: George Joseph <gjoseph@digium.com> +Date: Wed, 30 Jan 2019 12:25:55 -0700 +Subject: [PATCH] res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream + +apply_negotiated_sdp_stream was returning a "1" when no joint +capabilities were found on an outgoing call instead of a "-1". +This indicated to res_pjsip_session that the handler DID handle +the sdp when in fact it didn't. Without the appropriate setup, +a subsequent media frame coming in would have an invalid stream_num +and cause a seg fault when the stream was attempted to be retrieved. + +apply_negotiated_sdp_stream now returns the correct "-1" and any +media is now discarded before it reaches the core stream processing. + +ASTERISK-28620 +Reported by: Sotiris Ganouris + +Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f +--- + +diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c +index e2067cc..7f5a859 100644 +--- a/res/res_pjsip_sdp_rtp.c ++++ b/res/res_pjsip_sdp_rtp.c +@@ -1941,7 +1941,7 @@ + } + + if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) { +- return 1; ++ return -1; + } + + /* Set the channel uniqueid on the RTP instance now that it is becoming active */ |