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update to asterisk 11.8.0

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This commit is contained in:
sthen 2014-03-09 20:51:43 +00:00
parent dc23637394
commit 864e224cf3
3 changed files with 12 additions and 13 deletions
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7
telephony/asterisk/Makefile
View File

@ -1,13 +1,12 @@
# $OpenBSD: Makefile,v 1.191 2013/12/30 23:04:49 sthen Exp $
# $OpenBSD: Makefile,v 1.192 2014/03/09 20:51:43 sthen Exp $
SHARED_ONLY= Yes
COMMENT-main= open source multi-protocol PBX and telephony toolkit
VER= 11.7.0
VER= 11.8.0
DISTNAME= asterisk-${VER:S/beta/-beta/:S/rc/-rc/}
PKGNAME-main= asterisk-${VER}
REVISION-main= 0
CATEGORIES= telephony
@ -51,7 +50,7 @@ LIB_DEPENDS-main= audio/gsm \
textproc/libxml \
${MODGETTEXT_LIB_DEPENDS}
RUN_DEPENDS-main= ${MODGETTEXT_RUN_DEPENDS} \
telephony/asterisk-sounds/core-sounds/en,gsm \
telephony/asterisk-sounds/core-sounds/en,gsm>=1.4.25 \
telephony/asterisk-sounds/moh-opsound,wav
BUILD_DEPENDS= security/libsrtp>=1.4.4 # statically linked

4
telephony/asterisk/distinfo
View File

@ -1,2 +1,2 @@
SHA256 (asterisk-11.7.0.tar.gz) = T7KDuWRH9dhxYyoUBp81ZDyMNNCNbsX+3wrj3zDgw5c=
SIZE (asterisk-11.7.0.tar.gz) = 34779538
SHA256 (asterisk-11.8.0.tar.gz) = 2qKZzWgu8bxTtgHP3CRLkDzMn7xSwqz6FUWOPatv+kc=
SIZE (asterisk-11.8.0.tar.gz) = 34804028

14
telephony/asterisk/patches/patch-channels_chan_sip_c
View File

@ -1,11 +1,11 @@
$OpenBSD: patch-channels_chan_sip_c,v 1.12 2013/12/20 12:37:05 sthen Exp $
$OpenBSD: patch-channels_chan_sip_c,v 1.13 2014/03/09 20:51:43 sthen Exp $
https://issues.asterisk.org/jira/secure/attachment/46850/fax-deadlock-v2.patch-11.3.0
and add a bit more randomness for digest auth
--- channels/chan_sip.c.orig Fri Oct 25 17:05:55 2013
+++ channels/chan_sip.c Tue Dec 17 23:12:01 2013
@@ -8442,8 +8442,6 @@ static struct ast_frame *sip_read(struct ast_channel *
--- channels/chan_sip.c.orig Sat Mar 1 23:30:16 2014
+++ channels/chan_sip.c Mon Mar 3 23:22:16 2014
@@ -8444,8 +8444,6 @@ static struct ast_frame *sip_read(struct ast_channel *
ast_channel_unlock(ast);
if (ast_exists_extension(ast, target_context, "fax", 1,
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, NULL))) {
@ -14,7 +14,7 @@ and add a bit more randomness for digest auth
ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n", ast_channel_name(ast));
pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
if (ast_async_goto(ast, target_context, "fax", 1)) {
@@ -8452,10 +8450,10 @@ static struct ast_frame *sip_read(struct ast_channel *
@@ -8454,10 +8452,10 @@ static struct ast_frame *sip_read(struct ast_channel *
ast_frfree(fr);
fr = &ast_null_frame;
} else {
@ -27,7 +27,7 @@ and add a bit more randomness for digest auth
}
}
@@ -10722,6 +10720,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_r
@@ -10730,6 +10728,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_r
ast_channel_unlock(p->owner);
if (ast_exists_extension(p->owner, target_context, "fax", 1,
S_COR(ast_channel_caller(p->owner)->id.number.valid, ast_channel_caller(p->owner)->id.number.str, NULL))) {
@ -35,7 +35,7 @@ and add a bit more randomness for digest auth
ast_verb(2, "Redirecting '%s' to fax extension due to peer T.38 re-INVITE\n", ast_channel_name(p->owner));
pbx_builtin_setvar_helper(p->owner, "FAXEXTEN", ast_channel_exten(p->owner));
if (ast_async_goto(p->owner, target_context, "fax", 1)) {
@@ -16274,7 +16273,7 @@ static void build_route(struct sip_pvt *p, struct sip_
@@ -16313,7 +16312,7 @@ static void build_route(struct sip_pvt *p, struct sip_
static void build_nonce(struct sip_pvt *p, int forceupdate)
{
if (p->stalenonce || forceupdate || ast_strlen_zero(p->nonce)) {