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-rw-r--r--channels/chan_sip.c50
-rw-r--r--channels/sip/include/sip.h6
-rw-r--r--configs/sip.conf.sample12
-rw-r--r--main/features.c3
4 files changed, 61 insertions, 10 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 3f12e6e1b..99fb9b024 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -17751,6 +17751,8 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
ao2_t_ref(credentials, -1, "Unref peer auth for show");
}
ast_cli(fd, " Context : %s\n", peer->context);
+ ast_cli(fd, " Record On feature : %s\n", peer->record_on_feature);
+ ast_cli(fd, " Record Off feature : %s\n", peer->record_off_feature);
ast_cli(fd, " Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") );
ast_cli(fd, " Language : %s\n", peer->language);
ast_cli(fd, " Tonezone : %s\n", peer->zone[0] != '\0' ? peer->zone : "<Not set>");
@@ -18502,6 +18504,8 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " Allowed transports: %s\n", get_transport_list(default_transports));
ast_cli(a->fd, " Outbound transport: %s\n", sip_get_transport(default_primary_transport));
ast_cli(a->fd, " Context: %s\n", sip_cfg.default_context);
+ ast_cli(a->fd, " Record on feature: %s\n", sip_cfg.default_record_on_feature);
+ ast_cli(a->fd, " Record off feature: %s\n", sip_cfg.default_record_off_feature);
ast_cli(a->fd, " Force rport: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_NAT_FORCE_RPORT)));
ast_cli(a->fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF)));
ast_cli(a->fd, " Qualify: %d\n", default_qualify);
@@ -19204,15 +19208,13 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
return;
} else if (!ast_strlen_zero(c = sip_get_header(req, "Record"))) {
/* INFO messages generated by some phones to start/stop recording
- on phone calls.
- OEJ: I think this should be something that is enabled/disabled
- per device. I don't want incoming callers to record calls in my
- pbx.
- */
-
- struct ast_call_feature *feat;
+ * on phone calls.
+ */
+
+ struct ast_call_feature *feat = NULL;
int j;
struct ast_frame f = { AST_FRAME_DTMF, };
+ int suppress_warning = 0; /* Supress warning if the feature is blank */
if (!p->owner) { /* not a PBX call */
transmit_response(p, "481 Call leg/transaction does not exist", req);
@@ -19222,9 +19224,27 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
/* first, get the feature string, if it exists */
ast_rdlock_call_features();
- feat = ast_find_call_feature("automon");
+ if (p->relatedpeer) {
+ if (!strcasecmp(c, "on")) {
+ if (ast_strlen_zero(p->relatedpeer->record_on_feature)) {
+ suppress_warning = 1;
+ } else {
+ feat = ast_find_call_feature(p->relatedpeer->record_on_feature);
+ }
+ } else if (!strcasecmp(c, "off")) {
+ if (ast_strlen_zero(p->relatedpeer->record_on_feature)) {
+ suppress_warning = 1;
+ } else {
+ feat = ast_find_call_feature(p->relatedpeer->record_off_feature);
+ }
+ } else {
+ ast_log(LOG_ERROR, "Received INFO requesting to record with invalid value: %s\n", c);
+ }
+ }
if (!feat || ast_strlen_zero(feat->exten)) {
- ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
+ if (!suppress_warning) {
+ ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
+ }
/* 403 means that we don't support this feature, so don't request it again */
transmit_response(p, "403 Forbidden", req);
ast_unlock_call_features();
@@ -27666,6 +27686,8 @@ static void set_peer_defaults(struct sip_peer *peer)
ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
ast_copy_flags(&peer->flags[2], &global_flags[2], SIP_PAGE3_FLAGS_TO_COPY);
ast_string_field_set(peer, context, sip_cfg.default_context);
+ ast_string_field_set(peer, record_on_feature, sip_cfg.default_record_on_feature);
+ ast_string_field_set(peer, record_off_feature, sip_cfg.default_record_off_feature);
ast_string_field_set(peer, messagecontext, sip_cfg.messagecontext);
ast_string_field_set(peer, subscribecontext, sip_cfg.default_subscribecontext);
ast_string_field_set(peer, language, default_language);
@@ -27975,6 +27997,10 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
} else if (!strcasecmp(v->name, "context")) {
ast_string_field_set(peer, context, v->value);
ast_set_flag(&peer->flags[1], SIP_PAGE2_HAVEPEERCONTEXT);
+ } else if (!strcasecmp(v->name, "recordonfeature")) {
+ ast_string_field_set(peer, record_on_feature, v->value);
+ } else if (!strcasecmp(v->name, "recordofffeature")) {
+ ast_string_field_set(peer, record_off_feature, v->value);
} else if (!strcasecmp(v->name, "outofcall_message_context")) {
ast_string_field_set(peer, messagecontext, v->value);
} else if (!strcasecmp(v->name, "subscribecontext")) {
@@ -28734,6 +28760,8 @@ static int reload_config(enum channelreloadreason reason)
/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for devices */
ast_copy_string(sip_cfg.default_context, DEFAULT_CONTEXT, sizeof(sip_cfg.default_context));
+ ast_copy_string(sip_cfg.default_record_on_feature, DEFAULT_RECORD_FEATURE, sizeof(sip_cfg.default_record_on_feature));
+ ast_copy_string(sip_cfg.default_record_off_feature, DEFAULT_RECORD_FEATURE, sizeof(sip_cfg.default_record_off_feature));
sip_cfg.default_subscribecontext[0] = '\0';
sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
default_language[0] = '\0';
@@ -28801,6 +28829,10 @@ static int reload_config(enum channelreloadreason reason)
if (!strcasecmp(v->name, "context")) {
ast_copy_string(sip_cfg.default_context, v->value, sizeof(sip_cfg.default_context));
+ } else if (!strcasecmp(v->name, "recordonfeature")) {
+ ast_copy_string(sip_cfg.default_record_on_feature, v->value, sizeof(sip_cfg.default_record_on_feature));
+ } else if (!strcasecmp(v->name, "recordofffeature")) {
+ ast_copy_string(sip_cfg.default_record_off_feature, v->value, sizeof(sip_cfg.default_record_off_feature));
} else if (!strcasecmp(v->name, "subscribecontext")) {
ast_copy_string(sip_cfg.default_subscribecontext, v->value, sizeof(sip_cfg.default_subscribecontext));
} else if (!strcasecmp(v->name, "callcounter")) {
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 84f3af93b..ad9e7819f 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -34,6 +34,7 @@
#include "asterisk/astobj.h"
#include "asterisk/indications.h"
#include "asterisk/security_events.h"
+#include "asterisk/features.h"
#ifndef FALSE
#define FALSE 0
@@ -182,6 +183,7 @@
*/
/*@{*/
#define DEFAULT_CONTEXT "default" /*!< The default context for [general] section as well as devices */
+#define DEFAULT_RECORD_FEATURE "automon" /*!< The default feature specified for use with INFO */
#define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
#define DEFAULT_MOHSUGGEST ""
#define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
@@ -744,6 +746,8 @@ struct sip_settings {
struct sip_proxy outboundproxy; /*!< Outbound proxy */
char default_context[AST_MAX_CONTEXT];
char default_subscribecontext[AST_MAX_CONTEXT];
+ char default_record_on_feature[FEATURE_MAX_LEN];
+ char default_record_off_feature[FEATURE_MAX_LEN];
struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */
struct ast_format_cap *caps; /*!< Supported codecs */
int tcp_enabled;
@@ -1243,6 +1247,8 @@ struct sip_peer {
AST_STRING_FIELD(engine); /*!< RTP Engine to use */
AST_STRING_FIELD(unsolicited_mailbox); /*!< Mailbox to store received unsolicited MWI NOTIFY messages information in */
AST_STRING_FIELD(zone); /*!< Tonezone for this device */
+ AST_STRING_FIELD(record_on_feature); /*!< Feature to use when receiving INFO with record: on during a call */
+ AST_STRING_FIELD(record_off_feature); /*!< Feature to use when receiving INFO with record: off during a call */
);
struct sip_socket socket; /*!< Socket used for this peer */
enum sip_transport default_outbound_transport; /*!< Peer Registration may change the default outbound transport.
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 115b33709..358481143 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -143,6 +143,16 @@ allowoverlap=no ; Disable overlap dialing support. (Default is y
; In this case Realm will be based on request 'From'/'To' header
; and should match one of domain names.
; Otherwise default 'realm=...' will be used.
+;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
+ ; from an INFO message. Defaults to 'automon'. Works with
+ ; dynamic features. Feature must be usable on requesting
+ ; channel for it to work. Setting this value to a blank
+ ; will disable it.
+;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
+ ; from an INFO message. Defaults to 'automon'. Works with
+ ; dynamic features. Feature must be usable on requesting
+ ; channel for it to work. Setting this value to a blank
+ ; will disable it.
; With the current situation, you can do one of four things:
; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
@@ -1268,6 +1278,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;[grandstream1]
;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
+;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
+;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
diff --git a/main/features.c b/main/features.c
index ca6286bdc..7b595fb49 100644
--- a/main/features.c
+++ b/main/features.c
@@ -3015,7 +3015,8 @@ struct ast_call_feature *ast_find_call_feature(const char *name)
if (!strcasecmp(name, builtin_features[x].sname))
return &builtin_features[x];
}
- return NULL;
+
+ return find_dynamic_feature(name);
}
/*!
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-06 06:30:00 +0000