[svn-commits] mmichelson: trunk r394759 - in /trunk: channels/ include/asterisk/ res/ res/r...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jul 18 14:25:54 CDT 2013
Author: mmichelson
Date: Thu Jul 18 14:25:51 2013
New Revision: 394759
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=394759
Log:
Add a bunch of options from sip.conf to res_sip.conf
For a complete list of the options added, see the review linked
at the bottom of this commit message.
(closes issue ASTERISK-21506)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2671
Added:
trunk/res/res_sip/config_global.c
- copied unchanged from r394758, team/mmichelson/sip_options/res/res_sip/config_global.c
trunk/res/res_sip/config_system.c
- copied unchanged from r394758, team/mmichelson/sip_options/res/res_sip/config_system.c
trunk/res/res_sip/sip_global_headers.c
- copied unchanged from r394758, team/mmichelson/sip_options/res/res_sip/sip_global_headers.c
Modified:
trunk/channels/chan_gulp.c
trunk/include/asterisk/res_sip.h
trunk/res/res_sip.c
trunk/res/res_sip.exports.in
trunk/res/res_sip/config_transport.c
trunk/res/res_sip/include/res_sip_private.h
trunk/res/res_sip/sip_configuration.c
trunk/res/res_sip_caller_id.c
trunk/res/res_sip_mwi.c
trunk/res/res_sip_one_touch_record_info.c
trunk/res/res_sip_pubsub.c
trunk/res/res_sip_refer.c
trunk/res/res_sip_sdp_rtp.c
trunk/res/res_sip_session.c
Modified: trunk/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_gulp.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/channels/chan_gulp.c (original)
+++ trunk/channels/chan_gulp.c Thu Jul 18 14:25:51 2013
@@ -56,6 +56,7 @@
#include "asterisk/dsp.h"
#include "asterisk/stasis_endpoints.h"
#include "asterisk/stasis_channels.h"
+#include "asterisk/indications.h"
#include "asterisk/res_sip.h"
#include "asterisk/res_sip_session.h"
@@ -600,6 +601,18 @@
ast_channel_named_callgroups_set(chan, session->endpoint->named_callgroups);
ast_channel_named_pickupgroups_set(chan, session->endpoint->named_pickupgroups);
+
+ if (!ast_strlen_zero(session->endpoint->language)) {
+ ast_channel_language_set(chan, session->endpoint->language);
+ }
+
+ if (!ast_strlen_zero(session->endpoint->zone)) {
+ struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
+ if (!zone) {
+ ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
+ }
+ ast_channel_zone_set(chan, zone);
+ }
ast_endpoint_add_channel(session->endpoint->persistent, chan);
Modified: trunk/include/asterisk/res_sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_sip.h?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/include/asterisk/res_sip.h (original)
+++ trunk/include/asterisk/res_sip.h Thu Jul 18 14:25:51 2013
@@ -115,6 +115,10 @@
struct ast_sockaddr external_address;
/*! Transport state information */
struct ast_sip_transport_state *state;
+ /*! QOS DSCP TOS bits */
+ unsigned int tos;
+ /*! QOS COS value */
+ unsigned int cos;
};
/*!
@@ -323,6 +327,26 @@
AST_STRING_FIELD(external_media_address);
/*! Configured voicemail boxes for this endpoint. Used for MWI */
AST_STRING_FIELD(mailboxes);
+ /*! Configured RTP engine for this endpoint. */
+ AST_STRING_FIELD(rtp_engine);
+ /*! Configured tone zone for this endpoint. */
+ AST_STRING_FIELD(zone);
+ /*! Configured language for this endpoint. */
+ AST_STRING_FIELD(language);
+ /*! Feature to enact when one-touch recording INFO with Record: On is received */
+ AST_STRING_FIELD(recordonfeature);
+ /*! Feature to enact when one-touch recording INFO with Record: Off is received */
+ AST_STRING_FIELD(recordofffeature);
+ /*! SDP origin username */
+ AST_STRING_FIELD(sdpowner);
+ /*! SDP session name */
+ AST_STRING_FIELD(sdpsession);
+ /*! Default username to place in From header */
+ AST_STRING_FIELD(fromuser);
+ /*! Domain to place in From header */
+ AST_STRING_FIELD(fromdomain);
+ /*! Username to use when sending MWI NOTIFYs to this endpoint */
+ AST_STRING_FIELD(mwi_from);
);
/*! Identification information for this endpoint */
struct ast_party_id id;
@@ -408,6 +432,20 @@
struct ast_endpoint *persistent;
/*! The number of channels at which busy device state is returned */
unsigned int devicestate_busy_at;
+ /*! Determines if transfers (using REFER) are allowed by this endpoint */
+ unsigned int allowtransfer;
+ /*! DSCP TOS bits for audio streams */
+ unsigned int tos_audio;
+ /*! Priority for audio streams */
+ unsigned int cos_audio;
+ /*! DSCP TOS bits for video streams */
+ unsigned int tos_video;
+ /*! Priority for video streams */
+ unsigned int cos_video;
+ /*! Indicates if endpoint is allowed to initiate subscriptions */
+ unsigned int allowsubscribe;
+ /*! The minimum allowed expiration for subscriptions from endpoint */
+ unsigned int subminexpiry;
};
/*!
@@ -1057,13 +1095,15 @@
* \param endpoint Optional. If specified, the request will be created out-of-dialog
* to the endpoint.
* \param uri Optional. If specified, the request will be sent to this URI rather
+ * this value.
* than one configured for the endpoint.
* \param[out] tdata The newly-created request
* \retval 0 Success
* \retval -1 Failure
*/
int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
- struct ast_sip_endpoint *endpoint, const char *uri, pjsip_tx_data **tdata);
+ struct ast_sip_endpoint *endpoint, const char *uri,
+ pjsip_tx_data **tdata);
/*!
* \brief General purpose method for sending a SIP request
@@ -1308,4 +1348,12 @@
*/
void ast_sip_report_auth_challenge_sent(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pjsip_tx_data *tdata);
+void ast_sip_initialize_global_headers(void);
+void ast_sip_destroy_global_headers(void);
+
+int ast_sip_add_global_request_header(const char *name, const char *value, int replace);
+int ast_sip_add_global_response_header(const char *name, const char *value, int replace);
+
+int ast_sip_initialize_sorcery_global(struct ast_sorcery *sorcery);
+
#endif /* _RES_SIP_H */
Modified: trunk/res/res_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip.c (original)
+++ trunk/res/res_sip.c Thu Jul 18 14:25:51 2013
@@ -386,6 +386,96 @@
Gulp channel driver will return busy as the device state instead of in use.
</para></description>
</configOption>
+ <configOption name="tonezone">
+ <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
+ </configOption>
+ <configOption name="language">
+ <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
+ </configOption>
+ <configOption name="one_touch_recording" default="no">
+ <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
+ <see-also>
+ <ref type="configOption">recordonfeature</ref>
+ <ref type="configOption">recordofffeature</ref>
+ </see-also>
+ </configOption>
+ <configOption name="recordonfeature" default="automixmon">
+ <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
+ <description>
+ <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
+ feature will be enabled for the channel. The feature designated here can be any built-in
+ or dynamic feature defined in features.conf.</para>
+ <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
+ </description>
+ <see-also>
+ <ref type="configOption">one_touch_recording</ref>
+ <ref type="configOption">recordofffeature</ref>
+ </see-also>
+ </configOption>
+ <configOption name="recordofffeature" default="automixmon">
+ <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
+ <description>
+ <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
+ feature will be enabled for the channel. The feature designated here can be any built-in
+ or dynamic feature defined in features.conf.</para>
+ <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
+ </description>
+ <see-also>
+ <ref type="configOption">one_touch_recording</ref>
+ <ref type="configOption">recordonfeature</ref>
+ </see-also>
+ </configOption>
+ <configOption name="rtpengine" default="asterisk">
+ <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
+ </configOption>
+ <configOption name="allowtransfer" default="yes">
+ <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
+ </configOption>
+ <configOption name="sdpowner" default="-">
+ <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
+ </configOption>
+ <configOption name="sdpsession" default="Asterisk">
+ <synopsis>String used for the SDP session (s=) line.</synopsis>
+ </configOption>
+ <configOption name="tos_audio">
+ <synopsis>DSCP TOS bits for audio streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="tos_video">
+ <synopsis>DSCP TOS bits for video streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="cos_audio">
+ <synopsis>Priority for audio streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="cos_video">
+ <synopsis>Priority for video streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="allowsubscribe" default="no">
+ <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
+ </configOption>
+ <configOption name="subminexpiry" default="60">
+ <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
+ </configOption>
+ <configOption name="fromuser">
+ <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
+ </configOption>
+ <configOption name="mwifromuser">
+ <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
+ </configOption>
+ <configOption name="fromdomain">
+ <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
+ </configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>
@@ -674,6 +764,48 @@
</para></description>
</configOption>
</configObject>
+ <configObject name="system">
+ <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
+ <description><para>
+ The settings in this section are global. In addition to being global, the values will
+ not be re-evaluated when a reload is performed. This is because the values must be set
+ before the SIP stack is initialized. The only way to reset these values is to either
+ restart Asterisk, or unload res_sip.so and then load it again.
+ </para></description>
+ <configOption name="timert1" default="500">
+ <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
+ <description><para>
+ Timer T1 is the base for determining how long to wait before retransmitting
+ requests that receive no response when using an unreliable transport (e.g. UDP).
+ For more information on this timer, see RFC 3261, Section 17.1.1.1.
+ </para></description>
+ </configOption>
+ <configOption name="timerb" default="32000">
+ <synopsis>Set transaction timer B value (milliseconds).</synopsis>
+ <description><para>
+ Timer B determines the maximum amount of time to wait after sending an INVITE
+ request before terminating the transaction. It is recommended that this be set
+ to 64 * Timer T1, but it may be set higher if desired. For more information on
+ this timer, see RFC 3261, Section 17.1.1.1.
+ </para></description>
+ </configOption>
+ <configOption name="compactheaders" default="no">
+ <synopsis>Use the short forms of common SIP header names.</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="global">
+ <synopsis>Options that apply globally to all SIP communications</synopsis>
+ <description><para>
+ The settings in this section are global. Unlike options in the <literal>system</literal>
+ section, these options can be refreshed by performing a reload.
+ </para></description>
+ <configOption name="maxforwards" default="70">
+ <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
+ </configOption>
+ <configOption name="useragent" default="Asterisk <Asterisk Version>">
+ <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
+ </configOption>
+ </configObject>
</configFile>
</configInfo>
***/
@@ -865,7 +997,7 @@
return ast_pjsip_endpoint;
}
-static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const pj_str_t *target, pjsip_tpselector *selector)
+static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
{
pj_str_t tmp, local_addr;
pjsip_uri *uri;
@@ -874,7 +1006,7 @@
int local_port;
char uuid_str[AST_UUID_STR_LEN];
- if (!user) {
+ if (ast_strlen_zero(user)) {
RAII_VAR(struct ast_uuid *, uuid, ast_uuid_generate(), ast_free_ptr);
if (!uuid) {
return -1;
@@ -908,6 +1040,18 @@
/* If the host is IPv6 turn the transport into an IPv6 version */
if (pj_strchr(&sip_uri->host, ':') && type < PJSIP_TRANSPORT_START_OTHER) {
type = (pjsip_transport_type_e)(((int)type) + PJSIP_TRANSPORT_IPV6);
+ }
+
+ if (!ast_strlen_zero(domain)) {
+ from->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
+ from->slen = pj_ansi_snprintf(from->ptr, PJSIP_MAX_URL_SIZE,
+ "<%s:%s@%s%s%s>",
+ (pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
+ user,
+ domain,
+ (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
+ (type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
+ return 0;
}
/* Get the local bound address for the transport that will be used when communicating with the provided URI */
@@ -1000,7 +1144,7 @@
return NULL;
}
- if (sip_dialog_create_from(dlg->pool, &local_uri, NULL, &remote_uri, &selector)) {
+ if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
pjsip_dlg_terminate(dlg);
return NULL;
}
@@ -1127,7 +1271,8 @@
return -1;
}
- if (sip_dialog_create_from(pool, &from, NULL, &remote_uri, &selector)) {
+ if (sip_dialog_create_from(pool, &from, endpoint ? endpoint->fromuser : NULL,
+ endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
ast_log(LOG_ERROR, "Unable to create From header for %.*s request to endpoint %s\n",
(int) pj_strlen(&method->name), pj_strbuf(&method->name), ast_sorcery_object_get_id(endpoint));
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
@@ -1150,7 +1295,8 @@
}
int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
- struct ast_sip_endpoint *endpoint, const char *uri, pjsip_tx_data **tdata)
+ struct ast_sip_endpoint *endpoint, const char *uri,
+ pjsip_tx_data **tdata)
{
const pjsip_method *pmethod = get_pjsip_method(method);
@@ -1441,6 +1587,18 @@
}
return *servant_id == SIP_SERVANT_ID;
+}
+
+static void remove_request_headers(pjsip_endpoint *endpt)
+{
+ const pjsip_hdr *request_headers = pjsip_endpt_get_request_headers(endpt);
+ pjsip_hdr *iter = request_headers->next;
+
+ while (iter != request_headers) {
+ pjsip_hdr *to_erase = iter;
+ iter = iter->next;
+ pj_list_erase(to_erase);
+ }
}
static int load_module(void)
@@ -1480,9 +1638,20 @@
ast_log(LOG_ERROR, "Failed to create PJSIP endpoint structure. Aborting load\n");
goto error;
}
+
+ /* PJSIP will automatically try to add a Max-Forwards header. Since we want to control that,
+ * we need to stop PJSIP from doing it automatically
+ */
+ remove_request_headers(ast_pjsip_endpoint);
+
memory_pool = pj_pool_create(&caching_pool.factory, "SIP", 1024, 1024, NULL);
if (!memory_pool) {
ast_log(LOG_ERROR, "Failed to create memory pool for SIP. Aborting load\n");
+ goto error;
+ }
+
+ if (ast_sip_initialize_system()) {
+ ast_log(LOG_ERROR, "Failed to initialize SIP system configuration. Aborting load\n");
goto error;
}
@@ -1497,6 +1666,8 @@
goto error;
}
+ ast_sip_initialize_global_headers();
+
if (ast_res_sip_initialize_configuration()) {
ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
goto error;
@@ -1521,6 +1692,7 @@
error:
ast_sip_destroy_distributor();
ast_res_sip_destroy_configuration();
+ ast_sip_destroy_global_headers();
if (monitor_thread) {
stop_monitor_thread();
}
@@ -1564,6 +1736,7 @@
{
ast_sip_destroy_distributor();
ast_res_sip_destroy_configuration();
+ ast_sip_destroy_global_headers();
if (monitor_thread) {
stop_monitor_thread();
}
Modified: trunk/res/res_sip.exports.in
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip.exports.in?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip.exports.in (original)
+++ trunk/res/res_sip.exports.in Thu Jul 18 14:25:51 2013
@@ -60,6 +60,11 @@
LINKER_SYMBOL_PREFIXast_sip_report_auth_failed_challenge_response;
LINKER_SYMBOL_PREFIXast_sip_report_auth_success;
LINKER_SYMBOL_PREFIXast_sip_report_auth_challenge_sent;
+ LINKER_SYMBOL_PREFIXast_sip_initialize_global_headers;
+ LINKER_SYMBOL_PREFIXast_sip_destroy_global_headers;
+ LINKER_SYMBOL_PREFIXast_sip_add_global_request_header;
+ LINKER_SYMBOL_PREFIXast_sip_add_global_response_header;
+ LINKER_SYMBOL_PREFIXast_sip_initialize_sorcery_global;
local:
*;
};
Modified: trunk/res/res_sip/config_transport.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip/config_transport.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip/config_transport.c (original)
+++ trunk/res/res_sip/config_transport.c Thu Jul 18 14:25:51 2013
@@ -79,6 +79,18 @@
return transport;
}
+static void set_qos(struct ast_sip_transport *transport, pj_qos_params *qos)
+{
+ if (transport->tos) {
+ qos->flags |= PJ_QOS_PARAM_HAS_DSCP;
+ qos->dscp_val = transport->tos;
+ }
+ if (transport->cos) {
+ qos->flags |= PJ_QOS_PARAM_HAS_SO_PRIO;
+ qos->so_prio = transport->cos;
+ }
+}
+
/*! \brief Apply handler for transports */
static int transport_apply(const struct ast_sorcery *sorcery, void *obj)
{
@@ -135,12 +147,23 @@
} else if (transport->host.addr.sa_family == pj_AF_INET6()) {
res = pjsip_udp_transport_start6(ast_sip_get_pjsip_endpoint(), &transport->host.ipv6, NULL, transport->async_operations, &transport->state->transport);
}
+
+ if (res == PJ_SUCCESS && (transport->tos || transport->cos)) {
+ pj_sock_t sock;
+ pj_qos_params qos_params;
+
+ sock = pjsip_udp_transport_get_socket(transport->state->transport);
+ pj_sock_get_qos_params(sock, &qos_params);
+ set_qos(transport, &qos_params);
+ pj_sock_set_qos_params(sock, &qos_params);
+ }
} else if (transport->type == AST_TRANSPORT_TCP) {
pjsip_tcp_transport_cfg cfg;
pjsip_tcp_transport_cfg_default(&cfg, transport->host.addr.sa_family);
cfg.bind_addr = transport->host;
cfg.async_cnt = transport->async_operations;
+ set_qos(transport, &cfg.qos_params);
res = pjsip_tcp_transport_start3(ast_sip_get_pjsip_endpoint(), &cfg, &transport->state->factory);
} else if (transport->type == AST_TRANSPORT_TLS) {
@@ -148,9 +171,13 @@
transport->tls.cert_file = pj_str((char*)transport->cert_file);
transport->tls.privkey_file = pj_str((char*)transport->privkey_file);
transport->tls.password = pj_str((char*)transport->password);
+ set_qos(transport, &transport->tls.qos_params);
res = pjsip_tls_transport_start2(ast_sip_get_pjsip_endpoint(), &transport->tls, &transport->host, NULL, transport->async_operations, &transport->state->factory);
} else if ((transport->type == AST_TRANSPORT_WS) || (transport->type == AST_TRANSPORT_WSS)) {
+ if (transport->cos || transport->tos) {
+ ast_log(LOG_WARNING, "TOS and COS values ignored for websocket transport\n");
+ }
res = PJ_SUCCESS;
}
@@ -304,6 +331,8 @@
ast_sorcery_object_field_register_custom(sorcery, "transport", "method", "", transport_tls_method_handler, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sorcery, "transport", "cipher", "", transport_tls_cipher_handler, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sorcery, "transport", "localnet", "", transport_localnet_handler, NULL, 0, 0);
-
- return 0;
-}
+ ast_sorcery_object_field_register(sorcery, "transport", "tos", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_transport, tos));
+ ast_sorcery_object_field_register(sorcery, "transport", "cos", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_transport, cos));
+
+ return 0;
+}
Modified: trunk/res/res_sip/include/res_sip_private.h
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip/include/res_sip_private.h?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip/include/res_sip_private.h (original)
+++ trunk/res/res_sip/include/res_sip_private.h Thu Jul 18 14:25:51 2013
@@ -55,4 +55,20 @@
*/
int ast_sip_initialize_outbound_authentication(void);
+/*!
+ * \brief Initialize system configuration
+ *
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_sip_initialize_system(void);
+
+/*!
+ * \brief Initialize global configuration
+ *
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_sip_initialize_global(void);
+
#endif /* RES_SIP_PRIVATE_H_ */
Modified: trunk/res/res_sip/sip_configuration.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip/sip_configuration.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip/sip_configuration.c (original)
+++ trunk/res/res_sip/sip_configuration.c Thu Jul 18 14:25:51 2013
@@ -658,6 +658,24 @@
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "namedcallgroup", "", named_groups_handler, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "namedpickupgroup", "", named_groups_handler, NULL, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "devicestate_busy_at", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, devicestate_busy_at));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtpengine", "asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, rtp_engine));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "tonezone", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, zone));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "language", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, language));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "recordonfeature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, recordonfeature));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "recordofffeature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, recordofffeature));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allowtransfer", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allowtransfer));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdpowner", "-", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, sdpowner));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdpsession", "Asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, sdpsession));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "tos_audio", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, tos_audio));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "tos_video", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, tos_video));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "cos_audio", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, cos_audio));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "cos_video", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, cos_video));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allowsubscribe", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allowsubscribe));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subminexpiry", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, subminexpiry));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subminexpirey", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, subminexpiry));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "fromuser", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, fromuser));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "fromdomain", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, fromdomain));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "mwifromuser", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, mwi_from));
if (ast_sip_initialize_sorcery_transport(sip_sorcery)) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
@@ -691,6 +709,13 @@
if (ast_sip_initialize_sorcery_security(sip_sorcery)) {
ast_log(LOG_ERROR, "Failed to register SIP security support\n");
+ ast_sorcery_unref(sip_sorcery);
+ sip_sorcery = NULL;
+ return -1;
+ }
+
+ if (ast_sip_initialize_sorcery_global(sip_sorcery)) {
+ ast_log(LOG_ERROR, "Failed to register SIP Global support\n");
ast_sorcery_unref(sip_sorcery);
sip_sorcery = NULL;
return -1;
Modified: trunk/res/res_sip_caller_id.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_caller_id.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_caller_id.c (original)
+++ trunk/res/res_sip_caller_id.c Thu Jul 18 14:25:51 2013
@@ -398,7 +398,9 @@
pj_strdup2(pool, &id_name_addr->display, id->name.str);
}
- pj_strdup2(pool, &id_uri->user, id->number.str);
+ if (id->number.valid) {
+ pj_strdup2(pool, &id_uri->user, id->number.str);
+ }
}
/*!
Modified: trunk/res/res_sip_mwi.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_mwi.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_mwi.c (original)
+++ trunk/res/res_sip_mwi.c Thu Jul 18 14:25:51 2013
@@ -281,6 +281,14 @@
return 0;
}
+ if (!ast_strlen_zero(endpoint->mwi_from)) {
+ pjsip_fromto_hdr *from = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_FROM, NULL);
+ pjsip_name_addr *from_name_addr = (pjsip_name_addr *) from->uri;
+ pjsip_sip_uri *from_uri = pjsip_uri_get_uri(from_name_addr->uri);
+
+ pj_strdup2(tdata->pool, &from_uri->user, endpoint->mwi_from);
+ }
+
switch (state) {
case PJSIP_EVSUB_STATE_ACTIVE:
state_name = "active";
Modified: trunk/res/res_sip_one_touch_record_info.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_one_touch_record_info.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_one_touch_record_info.c (original)
+++ trunk/res/res_sip_one_touch_record_info.c Thu Jul 18 14:25:51 2013
@@ -51,12 +51,22 @@
pjsip_generic_string_hdr *record;
int feature_res;
char feature_code[AST_FEATURE_MAX_LEN];
+ const char *feature;
char *digit;
record = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &rec_str, NULL);
/* If we don't have Record header, we have nothing to do */
- if (!record || (pj_stricmp2(&record->hvalue, "on") && pj_stricmp2(&record->hvalue, "off"))) {
+ if (!record) {
+ return 0;
+ }
+
+ if (!pj_stricmp2(&record->hvalue, "on")) {
+ feature = session->endpoint->recordonfeature;
+ } else if (!pj_stricmp2(&record->hvalue, "off")) {
+ feature = session->endpoint->recordofffeature;
+ } else {
+ /* Don't send response because another module may handle this */
return 0;
}
@@ -66,13 +76,13 @@
}
/* Is this endpoint configured with One Touch Recording? */
- if (!session->endpoint->one_touch_recording) {
+ if (!session->endpoint->one_touch_recording || ast_strlen_zero(feature)) {
send_response(session, 403, rdata);
return 0;
}
ast_channel_lock(session->channel);
- feature_res = ast_get_builtin_feature(session->channel, "automixmon", feature_code, sizeof(feature_code));
+ feature_res = ast_get_feature(session->channel, feature, feature_code, sizeof(feature_code));
ast_channel_unlock(session->channel);
if (feature_res || ast_strlen_zero(feature_code)) {
Modified: trunk/res/res_sip_pubsub.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_pubsub.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_pubsub.c (original)
+++ trunk/res/res_sip_pubsub.c Thu Jul 18 14:25:51 2013
@@ -616,6 +616,7 @@
char accept[AST_SIP_MAX_ACCEPT][64];
pjsip_accept_hdr *accept_header;
pjsip_event_hdr *event_header;
+ pjsip_expires_hdr *expires_header;
struct ast_sip_subscription_handler *handler;
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
struct ast_sip_subscription *sub;
@@ -623,6 +624,21 @@
endpoint = ast_pjsip_rdata_get_endpoint(rdata);
ast_assert(endpoint != NULL);
+
+ if (!endpoint->allowsubscribe) {
+ ast_log(LOG_WARNING, "Subscriptions not permitted for endpoint %s.\n", ast_sorcery_object_get_id(endpoint));
+ pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 603, NULL, NULL, NULL);
+ return PJ_TRUE;
+ }
+
+ expires_header = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_EXPIRES, rdata->msg_info.msg->hdr.next);
+
+ if (expires_header && expires_header->ivalue < endpoint->subminexpiry) {
+ ast_log(LOG_WARNING, "Subscription expiration %d is too brief for endpoint %s. Minimum is %d\n",
+ expires_header->ivalue, ast_sorcery_object_get_id(endpoint), endpoint->subminexpiry);
+ pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 423, NULL, NULL, NULL);
+ return PJ_TRUE;
+ }
event_header = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_event_name, rdata->msg_info.msg->hdr.next);
if (!event_header) {
Modified: trunk/res/res_sip_refer.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_refer.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_refer.c (original)
+++ trunk/res/res_sip_refer.c Thu Jul 18 14:25:51 2013
@@ -732,6 +732,13 @@
pjsip_param *replaces;
int response;
+ if (!session->endpoint->allowtransfer) {
+ pjsip_dlg_respond(session->inv_session->dlg, rdata, 603, NULL, NULL, NULL);
+ ast_log(LOG_WARNING, "Endpoint %s transfer attempt blocked due to configuration\n",
+ ast_sorcery_object_get_id(session->endpoint));
+ return 0;
+ }
+
/* A Refer-To header is required */
if (!(refer_to = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL))) {
pjsip_dlg_respond(session->inv_session->dlg, rdata, 400, NULL, NULL, NULL);
Modified: trunk/res/res_sip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_sdp_rtp.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_sdp_rtp.c (original)
+++ trunk/res/res_sip_sdp_rtp.c Thu Jul 18 14:25:51 2013
@@ -108,7 +108,8 @@
{
struct ast_rtp_engine_ice *ice;
- if (!(session_media->rtp = ast_rtp_instance_new("asterisk", sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
+ if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->rtp_engine, sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
+ ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->rtp_engine);
return -1;
}
@@ -130,6 +131,16 @@
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
+ }
+
+ if (!strcmp(session_media->stream_type, STR_AUDIO) &&
+ (session->endpoint->tos_audio || session->endpoint->cos_video)) {
+ ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->tos_audio,
+ session->endpoint->cos_audio, "SIP RTP Audio");
+ } else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
+ (session->endpoint->tos_video || session->endpoint->cos_video)) {
+ ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->tos_video,
+ session->endpoint->cos_video, "SIP RTP Video");
}
return 0;
Modified: trunk/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_sip_session.c?view=diff&rev=394759&r1=394758&r2=394759
==============================================================================
--- trunk/res/res_sip_session.c (original)
+++ trunk/res/res_sip_session.c Thu Jul 18 14:25:51 2013
@@ -1684,7 +1684,6 @@
static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer)
{
RAII_VAR(struct ao2_iterator *, successful, NULL, ao2_iterator_cleanup);
- static const pj_str_t STR_ASTERISK = { "Asterisk", 8 };
static const pj_str_t STR_IN = { "IN", 2 };
static const pj_str_t STR_IP4 = { "IP4", 3 };
static const pj_str_t STR_IP6 = { "IP6", 3 };
@@ -1701,11 +1700,11 @@
local->origin.id = offer->origin.id;
}
- local->origin.user = STR_ASTERISK;
+ pj_strdup2(inv->pool, &local->origin.user, session->endpoint->sdpowner);
local->origin.net_type = STR_IN;
local->origin.addr_type = session->endpoint->rtp_ipv6 ? STR_IP6 : STR_IP4;
local->origin.addr = *pj_gethostname();
- local->name = local->origin.user;
+ pj_strdup2(inv->pool, &local->name, session->endpoint->sdpsession);
/* Now let the handlers add streams of various types, pjmedia will automatically reorder the media streams for us */
successful = ao2_callback_data(session->media, OBJ_MULTIPLE, add_sdp_streams, local, session);
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