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-rw-r--r--bridges/bridge_native_rtp.c6
-rw-r--r--configs/samples/pjsip.conf.sample19
-rw-r--r--include/asterisk/bridge_technology.h3
-rw-r--r--res/res_pjsip.c2
-rw-r--r--res/res_pjsip/pjsip_options.c4
-rw-r--r--res/res_pjsip_outbound_publish.c2
-rw-r--r--res/res_pjsip_outbound_registration.c13
7 files changed, 32 insertions, 17 deletions
diff --git a/bridges/bridge_native_rtp.c b/bridges/bridge_native_rtp.c
index a106d2d5f..a80ef4c5a 100644
--- a/bridges/bridge_native_rtp.c
+++ b/bridges/bridge_native_rtp.c
@@ -131,7 +131,7 @@ static void native_rtp_bridge_start(struct ast_bridge *bridge, struct ast_channe
{
struct ast_bridge_channel *bc0 = AST_LIST_FIRST(&bridge->channels);
struct ast_bridge_channel *bc1 = AST_LIST_LAST(&bridge->channels);
- enum ast_rtp_glue_result native_type;
+ enum ast_rtp_glue_result native_type = AST_RTP_GLUE_RESULT_FORBID;
struct ast_rtp_glue *glue0, *glue1;
RAII_VAR(struct ast_rtp_instance *, instance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ao2_cleanup);
@@ -147,7 +147,9 @@ static void native_rtp_bridge_start(struct ast_bridge *bridge, struct ast_channe
}
ast_channel_lock_both(bc0->chan, bc1->chan);
- native_type = native_rtp_bridge_get(bc0->chan, bc1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
+ if (!bc0->suspended && !bc1->suspended) {
+ native_type = native_rtp_bridge_get(bc0->chan, bc1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
+ }
switch (native_type) {
case AST_RTP_GLUE_RESULT_LOCAL:
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index bb2ad94f8..f66161329 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -640,7 +640,7 @@
;moh_suggest=default ; Default Music On Hold class (default: "default")
;outbound_auth= ; Authentication object used for outbound requests (default:
; "")
-;outbound_proxy= ; Proxy through which to send requests a full SIP URI
+;outbound_proxy= ; Proxy through which to send requests, a full SIP URI
; must be provided (default: "")
;rewrite_contact=no ; Allow Contact header to be rewritten with the source
; IP address port (default: "no")
@@ -865,8 +865,8 @@
;qualify_timeout=3.0 ; Qualify timeout in fractional seconds (default: "3.0")
;authenticate_qualify=no ; Authenticates a qualify request if needed
; (default: "no")
-;outbound_proxy= ; Outbound proxy used when sending OPTIONS request
- ; (default: "")
+;outbound_proxy= ; Proxy through which to send OPTIONS requests, a full SIP URI
+ ; must be provided (default: "")
;==========================SYSTEM SECTION OPTIONS=========================
@@ -898,11 +898,10 @@
;max_forwards=70 ; Value used in Max Forwards header for SIP requests
; (default: "70")
;type= ; Must be of type global (default: "")
-;user_agent=Asterisk PBX SVN-branch-12-r404375 ; Value used in User Agent
- ; header for SIP requests and
- ; Server header for SIP
- ; responses (default: "Asterisk
- ; PBX SVN-branch-12-r404375")
+;user_agent=Asterisk PBX ; Allows you to change the user agent string
+ ; The default user agent string also contains
+ ; the Asterisk version. If you don't want to
+ ; expose this, change the user_agent string.
;default_outbound_endpoint=default_outbound_endpoint ; Endpoint to use when
; sending an outbound
; request to a URI
@@ -1033,8 +1032,8 @@
;max_retries=10 ; Maximum number of registration attempts (default: "10")
;outbound_auth= ; Authentication object to be used for outbound registrations
; (default: "")
-;outbound_proxy= ; Outbound Proxy used to send registrations (default:
- ; "")
+;outbound_proxy= ; Proxy through which to send registrations, a full SIP URI
+ ; must be provided (default: "")
;retry_interval=60 ; Interval in seconds between retries if outbound
; registration is unsuccessful (default: "60")
;forbidden_retry_interval=0 ; Interval used when receiving a 403 Forbidden
diff --git a/include/asterisk/bridge_technology.h b/include/asterisk/bridge_technology.h
index 7f5d746f8..402b54e98 100644
--- a/include/asterisk/bridge_technology.h
+++ b/include/asterisk/bridge_technology.h
@@ -110,6 +110,9 @@ struct ast_bridge_technology {
*
* \note The bridge technology must tollerate a failed to join channel
* until it can be kicked from the bridge.
+ *
+ * \note A channel may be in a suspended state already when joining a bridge
+ * technology. The technology must handle this case.
*/
int (*join)(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel);
/*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 92dca7fb8..54a0a5f39 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -349,7 +349,7 @@
</description>
</configOption>
<configOption name="outbound_proxy">
- <synopsis>Proxy through which to send requests, a full SIP URI must be provided</synopsis>
+ <synopsis>Full SIP URI of the outbound proxy used to send requests</synopsis>
</configOption>
<configOption name="rewrite_contact">
<synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
diff --git a/res/res_pjsip/pjsip_options.c b/res/res_pjsip/pjsip_options.c
index d4159f508..d4ea911c9 100644
--- a/res/res_pjsip/pjsip_options.c
+++ b/res/res_pjsip/pjsip_options.c
@@ -1260,8 +1260,8 @@ int ast_sip_format_contact_ami(void *obj, void *arg, int flags)
if (!ast_strlen_zero(contact->call_id)) {
ast_str_append(&buf, 0, "CallID: %s\r\n", contact->call_id);
}
- ast_str_append(&buf, 0, "Status: %s\r\n", ast_sip_get_contact_status_label(status->status));
- if (status->status == UNKNOWN) {
+ ast_str_append(&buf, 0, "Status: %s\r\n", ast_sip_get_contact_status_label(status ? status->status : UNKNOWN));
+ if (!status || status->status == UNKNOWN) {
ast_str_append(&buf, 0, "RoundtripUsec: N/A\r\n");
} else {
ast_str_append(&buf, 0, "RoundtripUsec: %" PRId64 "\r\n", status->rtt);
diff --git a/res/res_pjsip_outbound_publish.c b/res/res_pjsip_outbound_publish.c
index 37f64481e..0273c6a68 100644
--- a/res/res_pjsip_outbound_publish.c
+++ b/res/res_pjsip_outbound_publish.c
@@ -68,7 +68,7 @@
</description>
</configOption>
<configOption name="outbound_proxy" default="">
- <synopsis>SIP URI of the outbound proxy used to send publishes</synopsis>
+ <synopsis>Full SIP URI of the outbound proxy used to send publishes</synopsis>
</configOption>
<configOption name="server_uri">
<synopsis>SIP URI of the server and entity to publish to</synopsis>
diff --git a/res/res_pjsip_outbound_registration.c b/res/res_pjsip_outbound_registration.c
index 122d5bb69..622df03f7 100644
--- a/res/res_pjsip_outbound_registration.c
+++ b/res/res_pjsip_outbound_registration.c
@@ -96,7 +96,7 @@
</description>
</configOption>
<configOption name="outbound_proxy" default="">
- <synopsis>Outbound Proxy used to send registrations</synopsis>
+ <synopsis>Full SIP URI of the outbound proxy used to send registrations</synopsis>
</configOption>
<configOption name="retry_interval" default="60">
<synopsis>Interval in seconds between retries if outbound registration is unsuccessful</synopsis>
@@ -1210,6 +1210,17 @@ static int sip_outbound_registration_regc_alloc(void *data)
return -1;
}
+ if (!ast_strlen_zero(registration->outbound_proxy)) {
+ pj_strdup2_with_null(pool, &tmp, registration->outbound_proxy);
+ uri = pjsip_parse_uri(pool, tmp.ptr, tmp.slen, 0);
+ if (!uri) {
+ ast_log(LOG_ERROR, "Invalid outbound proxy URI '%s' specified on outbound registration '%s'\n",
+ registration->outbound_proxy, ast_sorcery_object_get_id(registration));
+ pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
+ return -1;
+ }
+ }
+
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-28 19:03:39 +0000