Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."

CHANGES

@@ -141,6 +141,10 @@ res_pjsip
    added to both transport and subscription_persistence, an alembic upgrade
    should be run to bring the database tables up to date.
 
+ * A new option, allow_overlap, has been added to endpoints which allows
+   overlap dialing functionality to be enabled or disabled. The option defaults
+   to enabled.
+
 res_pjsip_transport_websocket
 ------------------
  * Removed non-secure websocket support.  Firefox and Chrome have not allowed

configs/samples/pjsip.conf.sample

@@ -595,6 +595,7 @@
                 ; "yes")
 ;aggregate_mwi=yes      ;  (default: "yes")
 ;allow= ; Media Codec s to allow (default: "")
+;allow_overlap=yes ; Enable RFC3578 overlap dialing support. (default: "yes")
 ;aors=  ; AoR s to be used with the endpoint (default: "")
 ;auth=  ; Authentication Object s associated with the endpoint (default: "")
 ;callerid=      ; CallerID information for the endpoint (default: "")

contrib/ast-db-manage/config/versions/8fce4c573e15_add_pjsip_allow_overlap.py

@@ -0,0 +1,31 @@
+"""add pjsip allow_overlap
+
+Revision ID: 8fce4c573e15
+Revises: f638dbe2eb23
+Create Date: 2017-03-21 15:14:27.612945
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '8fce4c573e15'
+down_revision = 'f638dbe2eb23'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+    ############################# Enums ##############################
+
+    # yesno_values have already been created, so use postgres enum object
+    # type to get around "already created" issue - works okay with mysql
+    yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+    op.add_column('ps_endpoints', sa.Column('allow_overlap', yesno_values))
+
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'allow_overlap')

include/asterisk/res_pjsip.h

@@ -765,6 +765,8 @@ struct ast_sip_endpoint {
 	unsigned int preferred_codec_only;
 	/*! Do we allow an asymmetric RTP codec? */
 	unsigned int asymmetric_rtp_codec;
+	/*! Do we allow overlap dialling? */
+	unsigned int allow_overlap;
 };
 
 /*! URI parameter for symmetric transport */

res/res_pjsip.c

@@ -100,6 +100,9 @@
 				<configOption name="allow">
 					<synopsis>Media Codec(s) to allow</synopsis>
 				</configOption>
+				<configOption name="allow_overlap" default="yes">
+					<synopsis>Enable RFC3578 overlap dialing support.</synopsis>
+				</configOption>
 				<configOption name="aors">
 					<synopsis>AoR(s) to be used with the endpoint</synopsis>
 					<description><para>
@@ -2134,6 +2137,9 @@
 				<parameter name="SubscribeContext">
 					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='subscribe_context']/synopsis/node())"/></para>
 				</parameter>
+				<parameter name="Allowoverlap">
+					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_overlap']/synopsis/node())"/></para>
+				</parameter>
 			</syntax>
 		</managerEventInstance>
 	</managerEvent>

res/res_pjsip/pjsip_configuration.c

@@ -1938,6 +1938,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "preferred_codec_only", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, preferred_codec_only));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "asymmetric_rtp_codec", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, asymmetric_rtp_codec));
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtcp_mux", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtcp_mux));
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow_overlap", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allow_overlap));
 
 	if (ast_sip_initialize_sorcery_transport()) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");

res/res_pjsip_session.c

@@ -1986,10 +1986,17 @@ static enum sip_get_destination_result get_destination(struct ast_sip_session *s
 
 		return SIP_GET_DEST_EXTEN_FOUND;
 	}
-	/* XXX In reality, we'll likely have further options so that partial matches
-	 * can be indicated here, but for getting something up and running, we're going
-	 * to return a "not exists" error here.
+
+	/*
+	 * Check for partial match via overlap dialling (if enabled)
 	 */
+	if (session->endpoint->allow_overlap && (
+		!strncmp(session->exten, pickupexten, strlen(session->exten)) ||
+		ast_canmatch_extension(NULL, session->endpoint->context, session->exten, 1, NULL))) {
+		/* Overlap partial match */
+		return SIP_GET_DEST_EXTEN_PARTIAL;
+	}
+
 	return SIP_GET_DEST_EXTEN_NOT_FOUND;
 }
 
@@ -2106,8 +2113,17 @@ static int new_invite(void *data)
 			pjsip_inv_terminate(invite->session->inv_session, 416, PJ_TRUE);
 		}
 		goto end;
-	case SIP_GET_DEST_EXTEN_NOT_FOUND:
 	case SIP_GET_DEST_EXTEN_PARTIAL:
+		ast_debug(1, "Call from '%s' (%s:%s:%d) to extension '%s' - partial match\n", ast_sorcery_object_get_id(invite->session->endpoint),
+			invite->rdata->tp_info.transport->type_name, invite->rdata->pkt_info.src_name, invite->rdata->pkt_info.src_port, invite->session->exten);
+
+		if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 484, NULL, NULL, &tdata) == PJ_SUCCESS) {
+			ast_sip_session_send_response(invite->session, tdata);
+		} else  {
+			pjsip_inv_terminate(invite->session->inv_session, 484, PJ_TRUE);
+		}
+		goto end;
+	case SIP_GET_DEST_EXTEN_NOT_FOUND:
 	default:
 		ast_log(LOG_NOTICE, "Call from '%s' (%s:%s:%d) to extension '%s' rejected because extension not found in context '%s'.\n",
 			ast_sorcery_object_get_id(invite->session->endpoint), invite->rdata->tp_info.transport->type_name, invite->rdata->pkt_info.src_name,