diff --git a/CHANGES b/CHANGES
index 08c9185b6a5ef807c3c1516fc58d100e5aa91ddc..2304d09d761b98e56c665ca279f242ab57adec9b 100644
--- a/CHANGES
+++ b/CHANGES
@@ -57,6 +57,10 @@ 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
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 82da311a0c9b38462dda3eea27d33edb7315b6f7..82cfc09ae7d9173b606bfd2c0e4e897c9f3fb879 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/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: "")
diff --git a/contrib/ast-db-manage/config/versions/8fce4c573e15_add_pjsip_allow_overlap.py b/contrib/ast-db-manage/config/versions/8fce4c573e15_add_pjsip_allow_overlap.py
new file mode 100644
index 0000000000000000000000000000000000000000..24057ecc878e451490f0bb4a89a59d1bb46a32b1
--- /dev/null
+++ b/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')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 05a3eea44f841c87f394f132c744aecab8d9fece..59122b987399e447b9a6f1e67eae21a53e1d4194 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -759,6 +759,8 @@ struct ast_sip_endpoint {
unsigned int asymmetric_rtp_codec;
/*! Use RTCP-MUX */
unsigned int rtcp_mux;
+ /*! Do we allow overlap dialling? */
+ unsigned int allow_overlap;
};
/*! URI parameter for symmetric transport */
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 7b10f47f63b17e2b5b50b9a1757c27c0978705d8..fc49856570e8a32eb946f740d3ac717e06db5e92 100644
--- a/res/res_pjsip.c
+++ b/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>
@@ -2122,6 +2125,9 @@
<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>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index eb8e1971283e3174e17300f742d798be282acb76..511ea41c14bdb320e44b86b0f9e8b8effae07b73 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1939,6 +1939,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_user", "", contact_user_handler, contact_user_to_str, NULL, 0, 0);
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, 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");
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index efdce8aa2656630af6fb3a610255a623b561eb2d..53841c44af1e0506b49325a30bda3b70d4bbcdc2 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1984,10 +1984,17 @@ 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;
}
@@ -2104,8 +2111,17 @@ 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,