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-rw-r--r--CHANGES7
-rw-r--r--channels/chan_sip.c20
-rw-r--r--channels/sip/include/sip.h1
-rw-r--r--configs/samples/sip.conf.sample4
4 files changed, 30 insertions, 2 deletions
diff --git a/CHANGES b/CHANGES
index 044605cc5..8c476e433 100644
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,13 @@
--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
------------------------------------------------------------------------------
+chan_sip
+------------------
+ * The websockets_enabled option has been added to the general section of
+ sip.conf. The option is enabled by default to match the previous behavior.
+ The option should be disabled when using res_pjsip_transport_websockets to
+ ensure chan_sip will not conflict with PJSIP websockets.
+
Dialplan Functions
------------------
* The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f28296627..acb7d535f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -31261,6 +31261,7 @@ static int reload_config(enum channelreloadreason reason)
int bindport = 0;
int acl_change_subscription_needed = 0;
int min_subexpiry_set = 0, max_subexpiry_set = 0;
+ int websocket_was_enabled = sip_cfg.websocket_enabled;
run_start = time(0);
ast_unload_realtime("sipregs");
@@ -32047,6 +32048,8 @@ static int reload_config(enum channelreloadreason reason)
ast_log(LOG_WARNING, "'%s' is not a valid websocket_write_timeout value at line %d. Using default '%d'.\n", v->value, v->lineno, AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT);
sip_cfg.websocket_write_timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
}
+ } else if (!strcasecmp(v->name, "websocket_enabled")) {
+ sip_cfg.websocket_enabled = ast_true(v->value);
}
}
@@ -32392,6 +32395,15 @@ static int reload_config(enum channelreloadreason reason)
notify_types = NULL;
}
+ /* If the module is loading it's not time to enable websockets yet. */
+ if (reason != CHANNEL_MODULE_LOAD && websocket_was_enabled != sip_cfg.websocket_enabled) {
+ if (sip_cfg.websocket_enabled) {
+ ast_websocket_add_protocol("sip", sip_websocket_callback);
+ } else {
+ ast_websocket_remove_protocol("sip", sip_websocket_callback);
+ }
+ }
+
run_end = time(0);
ast_debug(4, "SIP reload_config done...Runtime= %d sec\n", (int)(run_end-run_start));
@@ -34573,7 +34585,9 @@ static int load_module(void)
sip_register_tests();
network_change_stasis_subscribe();
- ast_websocket_add_protocol("sip", sip_websocket_callback);
+ if (sip_cfg.websocket_enabled) {
+ ast_websocket_add_protocol("sip", sip_websocket_callback);
+ }
return AST_MODULE_LOAD_SUCCESS;
}
@@ -34588,7 +34602,9 @@ static int unload_module(void)
ast_sip_api_provider_unregister();
- ast_websocket_remove_protocol("sip", sip_websocket_callback);
+ if (sip_cfg.websocket_enabled) {
+ ast_websocket_remove_protocol("sip", sip_websocket_callback);
+ }
network_change_stasis_unsubscribe();
acl_change_event_stasis_unsubscribe();
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 3ed3e8a33..82f208c77 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -774,6 +774,7 @@ struct sip_settings {
int tcp_enabled;
int default_max_forwards; /*!< Default max forwards (SIP Anti-loop) */
int websocket_write_timeout; /*!< Socket write timeout for websocket transports, in ms */
+ int websocket_enabled; /*!< Are websockets enabled? */
};
struct ast_websocket;
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index 44d2d4352..a24ab30a6 100644
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -229,6 +229,10 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
; unauthenticated sessions that will be allowed
; to connect at any given time. (default: 100)
+;websocket_enabled = true ; Set to false to prevent chan_sip from listening to websockets. This
+ ; is neeeded when using chan_sip and res_pjsip_transport_websockets on
+ ; the same system.
+
;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
; This value may need to be adjusted for connections where
; Asterisk must write a substantial amount of data and the