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-rw-r--r--Makefile20
-rw-r--r--apps/app_osplookup.c6
-rw-r--r--apps/app_voicemail.c74
-rw-r--r--cdr/cdr_adaptive_odbc.c1
-rw-r--r--cdr/cdr_odbc.c1
-rw-r--r--cel/cel_odbc.c1
-rw-r--r--channels/chan_sip.c14
-rw-r--r--funcs/func_odbc.c1
-rw-r--r--include/asterisk/astosp.h31
-rw-r--r--res/res_config_odbc.c1
-rw-r--r--res/res_http_post.c2
-rw-r--r--res/res_pjsip_session.c31
12 files changed, 115 insertions, 68 deletions
diff --git a/Makefile b/Makefile
index 4e6072e3a..e624abf77 100644
--- a/Makefile
+++ b/Makefile
@@ -239,7 +239,9 @@ ifeq ($(OSARCH),SunOS)
_ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2 -D__EXTENSIONS__
endif
-ifneq ($(GREP),:)
+ifeq ($(GREP),)
+else ifeq ($(GREP),:)
+else
ASTERISKVERSION:=$(shell GREP=$(GREP) AWK=$(AWK) GIT=$(GIT) build_tools/make_version .)
endif
ifneq ($(AWK),)
@@ -467,7 +469,9 @@ endif
$(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/rest-api" ; \
done
-ifneq ($(GREP),:)
+ifeq ($(GREP),)
+else ifeq ($(GREP),:)
+else
XML_core_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
endif
@@ -486,7 +490,9 @@ doc/core-en_US.xml: makeopts .lastclean $(XML_core_en_US)
@echo
@echo "</docs>" >> $@
-ifneq ($(GREP),:)
+ifeq ($(GREP),)
+else ifeq ($(GREP),:)
+else
XML_full_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
endif
@@ -630,7 +636,9 @@ oldmodcheck:
fi
ld-cache-update:
-ifneq ($(LDCONFIG),:)
+ifeq ($(LDCONFIG),)
+else ifeq ($(LDCONFIG),:)
+else
ifeq ($(DESTDIR),) # DESTDIR means binary archive creation; ldconfig should be run on postinst
@if [ $$(id -u) -eq 0 ] ; then \
$(LDCONFIG) "$(ASTLIBDIR)/" ; \
@@ -984,7 +992,9 @@ ifeq ($(HAVE_DAHDI),1)
rm -f $(DESTDIR)$(DAHDI_UDEV_HOOK_DIR)/40-asterisk
endif
$(MAKE) -C sounds uninstall
-ifneq ($(LDCONFIG),:)
+ifeq ($(LDCONFIG),)
+else ifeq ($(LDCONFIG),:)
+else
$(LDCONFIG) "$(ASTLIBDIR)/" || :
endif
diff --git a/apps/app_osplookup.c b/apps/app_osplookup.c
index c42e1873e..78f573a6c 100644
--- a/apps/app_osplookup.c
+++ b/apps/app_osplookup.c
@@ -50,7 +50,6 @@
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/cli.h"
-#include "asterisk/astosp.h"
/*** DOCUMENTATION
<application name="OSPAuth" language="en_US">
@@ -436,6 +435,11 @@
</application>
***/
+/* OSP Return statuses */
+#define AST_OSP_SUCCESS ((char*)"SUCCESS") /* Return status, success */
+#define AST_OSP_FAILED ((char*)"FAILED") /* Return status, failed */
+#define AST_OSP_ERROR ((char*)"ERROR") /* Return status, error */
+
/* OSP Buffer Sizes */
#define OSP_SIZE_INTSTR ((unsigned int)16) /* OSP signed/unsigned int string buffer size */
#define OSP_SIZE_NORSTR ((unsigned int)256) /* OSP normal string buffer size */
diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index 72e3e59c1..1e7d09128 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -6089,22 +6089,33 @@ static int has_voicemail(const char *mailbox, const char *folder)
return 0;
}
-
+/*!
+ * \brief Check the given mailbox's message count.
+ * \param mailbox The @ delimited string for user@context. If no context is found, uses 'default' for the context.
+ * \param urgentmsgs urgent message count.
+ * \param newmsgs new message count.
+ * \param oldmsgs old message count pointer
+ * \return -1 if error occurred, 0 otherwise.
+ */
static int inboxcount2(const char *mailbox, int *urgentmsgs, int *newmsgs, int *oldmsgs)
{
char tmp[256];
char *context;
/* If no mailbox, return immediately */
- if (ast_strlen_zero(mailbox))
+ if (ast_strlen_zero(mailbox)) {
return 0;
+ }
- if (newmsgs)
+ if (newmsgs) {
*newmsgs = 0;
- if (oldmsgs)
+ }
+ if (oldmsgs) {
*oldmsgs = 0;
- if (urgentmsgs)
+ }
+ if (urgentmsgs) {
*urgentmsgs = 0;
+ }
if (strchr(mailbox, ',')) {
int tmpnew, tmpold, tmpurgent;
@@ -6114,15 +6125,18 @@ static int inboxcount2(const char *mailbox, int *urgentmsgs, int *newmsgs, int *
mb = tmp;
while ((cur = strsep(&mb, ", "))) {
if (!ast_strlen_zero(cur)) {
- if (inboxcount2(cur, urgentmsgs ? &tmpurgent : NULL, newmsgs ? &tmpnew : NULL, oldmsgs ? &tmpold : NULL))
+ if (inboxcount2(cur, urgentmsgs ? &tmpurgent : NULL, newmsgs ? &tmpnew : NULL, oldmsgs ? &tmpold : NULL)) {
return -1;
- else {
- if (newmsgs)
+ } else {
+ if (newmsgs) {
*newmsgs += tmpnew;
- if (oldmsgs)
+ }
+ if (oldmsgs) {
*oldmsgs += tmpold;
- if (urgentmsgs)
+ }
+ if (urgentmsgs) {
*urgentmsgs += tmpurgent;
+ }
}
}
}
@@ -6131,17 +6145,21 @@ static int inboxcount2(const char *mailbox, int *urgentmsgs, int *newmsgs, int *
ast_copy_string(tmp, mailbox, sizeof(tmp));
- if ((context = strchr(tmp, '@')))
+ if ((context = strchr(tmp, '@'))) {
*context++ = '\0';
- else
+ } else {
context = "default";
+ }
- if (newmsgs)
+ if (newmsgs) {
*newmsgs = __has_voicemail(context, tmp, "INBOX", 0);
- if (oldmsgs)
+ }
+ if (oldmsgs) {
*oldmsgs = __has_voicemail(context, tmp, "Old", 0);
- if (urgentmsgs)
+ }
+ if (urgentmsgs) {
*urgentmsgs = __has_voicemail(context, tmp, "Urgent", 0);
+ }
return 0;
}
@@ -13241,7 +13259,10 @@ static void stop_poll_thread(void)
poll_thread = AST_PTHREADT_NULL;
}
-/*! \brief Append vmu info string into given astman with event_name. */
+/*!
+ * \brief Append vmu info string into given astman with event_name.
+ * \return 0 failed. 1 otherwise.
+*/
static int append_vmu_info_astman(
struct mansession *s,
struct ast_vm_user *vmu,
@@ -13251,14 +13272,33 @@ static int append_vmu_info_astman(
{
int new;
int old;
+ char *mailbox;
+ int ret;
if((s == NULL) || (vmu == NULL) || (event_name == NULL) || (actionid == NULL)) {
ast_log(LOG_ERROR, "Wrong input parameter.");
return 0;
}
+ /* create mailbox string */
+ if (!ast_strlen_zero(vmu->context)) {
+ ret = ast_asprintf(&mailbox, "%s@%s", vmu->mailbox, vmu->context);
+ } else {
+ ret = ast_asprintf(&mailbox, "%s", vmu->mailbox);
+ }
+ if (ret == -1) {
+ ast_log(LOG_ERROR, "Could not create mailbox string. err[%s]\n", strerror(errno));
+ return 0;
+ }
+
/* get mailbox count */
- inboxcount(vmu->mailbox, &new, &old);
+ ret = inboxcount(mailbox, &new, &old);
+ ast_free(mailbox);
+ if (ret == -1) {
+ ast_log(LOG_ERROR, "Could not get mailbox count. user[%s], context[%s]\n",
+ vmu->mailbox ?: "", vmu->context ?: "");
+ return 0;
+ }
astman_append(s,
"Event: %s\r\n"
diff --git a/cdr/cdr_adaptive_odbc.c b/cdr/cdr_adaptive_odbc.c
index d4da850ce..e04b9fec8 100644
--- a/cdr/cdr_adaptive_odbc.c
+++ b/cdr/cdr_adaptive_odbc.c
@@ -35,6 +35,7 @@
/*** MODULEINFO
<depend>res_odbc</depend>
+ <depend>generic_odbc</depend>
<support_level>core</support_level>
***/
diff --git a/cdr/cdr_odbc.c b/cdr/cdr_odbc.c
index b44bbf52a..efa68c1a9 100644
--- a/cdr/cdr_odbc.c
+++ b/cdr/cdr_odbc.c
@@ -39,6 +39,7 @@
/*** MODULEINFO
<depend>res_odbc</depend>
+ <depend>generic_odbc</depend>
<support_level>extended</support_level>
***/
diff --git a/cel/cel_odbc.c b/cel/cel_odbc.c
index 05c1095cb..b4967b647 100644
--- a/cel/cel_odbc.c
+++ b/cel/cel_odbc.c
@@ -28,6 +28,7 @@
/*** MODULEINFO
<depend>res_odbc</depend>
+ <depend>generic_odbc</depend>
<support_level>core</support_level>
***/
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 85701634c..f0cc2a6bd 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10964,22 +10964,25 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (portno != -1 || vportno != -1 || tportno != -1) {
/* We are now ready to change the sip session and RTP structures with the offered codecs, since
they are acceptable */
+ unsigned int framing;
ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(p->jointcaps, newjointcapability, AST_MEDIA_TYPE_UNKNOWN); /* Our joint codec profile for this call */
ast_format_cap_remove_by_type(p->peercaps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(p->peercaps, newpeercapability, AST_MEDIA_TYPE_UNKNOWN); /* The other side's capability in latest offer */
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
+ tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
+ framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
/* respond with single most preferred joint codec, limiting the other side's choice */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
- unsigned int framing;
-
- tmp_fmt = ast_format_cap_get_format(p->jointcaps, 0);
- framing = ast_format_cap_get_format_framing(p->jointcaps, tmp_fmt);
ast_format_cap_remove_by_type(p->jointcaps, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append(p->jointcaps, tmp_fmt, framing);
- ao2_ref(tmp_fmt, -1);
}
+ if (!ast_rtp_codecs_get_framing(&newaudiortp)) {
+ /* Peer did not force us to use a specific framing, so use our own */
+ ast_rtp_codecs_set_framing(&newaudiortp, framing);
+ }
+ ao2_ref(tmp_fmt, -1);
}
/* Setup audio address and port */
@@ -11488,6 +11491,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
if (framing && p->autoframing) {
ast_debug(1, "Setting framing to %ld\n", framing);
ast_format_cap_set_framing(p->caps, framing);
+ ast_rtp_codecs_set_framing(newaudiortp, framing);
}
found = TRUE;
} else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
diff --git a/funcs/func_odbc.c b/funcs/func_odbc.c
index 0f6b65e7d..0d053f614 100644
--- a/funcs/func_odbc.c
+++ b/funcs/func_odbc.c
@@ -29,6 +29,7 @@
/*** MODULEINFO
<depend>res_odbc</depend>
+ <depend>generic_odbc</depend>
<support_level>core</support_level>
***/
diff --git a/include/asterisk/astosp.h b/include/asterisk/astosp.h
deleted file mode 100644
index 4faa2b4e7..000000000
--- a/include/asterisk/astosp.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*!
- * \file
- * \brief Open Settlement Protocol (OSP)
- */
-
-#ifndef _ASTERISK_OSP_H
-#define _ASTERISK_OSP_H
-
-#define AST_OSP_SUCCESS ((char*)"SUCCESS") /* Return status, success */
-#define AST_OSP_FAILED ((char*)"FAILED") /* Return status, failed */
-#define AST_OSP_ERROR ((char*)"ERROR") /* Return status, error */
-
-#endif /* _ASTERISK_OSP_H */
diff --git a/res/res_config_odbc.c b/res/res_config_odbc.c
index 186e89dcd..5d6c2c60a 100644
--- a/res/res_config_odbc.c
+++ b/res/res_config_odbc.c
@@ -30,6 +30,7 @@
/*** MODULEINFO
<depend>res_odbc</depend>
+ <depend>generic_odbc</depend>
<support_level>core</support_level>
***/
diff --git a/res/res_http_post.c b/res/res_http_post.c
index 44bb8ee9d..f36b772d4 100644
--- a/res/res_http_post.c
+++ b/res/res_http_post.c
@@ -36,7 +36,7 @@
#include <sys/stat.h>
#include <fcntl.h>
#include <gmime/gmime.h>
-#if defined (__OpenBSD__) || defined(__FreeBSD__) || defined(__Darwin__)
+#if defined (__OpenBSD__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__Darwin__)
#include <libgen.h>
#endif
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index fcd190bcb..f25201731 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -876,15 +876,30 @@ static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_
struct ast_stream_topology *topology;
unsigned int changed = 0;
- /* This situation can legitimately happen when an SDP is received in a
- * 183 Session Progress message. In that case, everything's been done
- * by the time this function is called and there are no more pending
- * streams.
- */
if (!session->pending_media_state->topology) {
- ast_debug(1, "Pending topology was NULL for channel '%s'\n",
- session->channel ? ast_channel_name(session->channel) : "unknown");
- return 0;
+ if (session->active_media_state->topology) {
+ /*
+ * This happens when we have negotiated media after receiving a 183,
+ * and we're now receiving a 200 with a new SDP. In this case, there
+ * is active_media_state, but the pending_media_state has been reset.
+ */
+ struct ast_sip_session_media_state *active_media_state_clone;
+
+ active_media_state_clone =
+ ast_sip_session_media_state_clone(session->active_media_state);
+ if (!active_media_state_clone) {
+ ast_log(LOG_WARNING, "Unable to clone active media state for channel '%s'\n",
+ session->channel ? ast_channel_name(session->channel) : "unknown");
+ return -1;
+ }
+
+ ast_sip_session_media_state_free(session->pending_media_state);
+ session->pending_media_state = active_media_state_clone;
+ } else {
+ ast_log(LOG_WARNING, "No pending or active media state for channel '%s'\n",
+ session->channel ? ast_channel_name(session->channel) : "unknown");
+ return -1;
+ }
}
/* If we're handling negotiated streams, then we should already have set
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-02 21:35:31 +0000