diff options
author | Joshua Colp <jcolp@digium.com> | 2017-03-22 17:08:08 -0500 |
---|---|---|
committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-03-22 17:08:08 -0500 |
commit | c1ab8ca74cec45730107cca3ed47fc61460365e4 (patch) | |
tree | ed493e880d480741455c2b1fc3e9fecc8556a2e7 /res | |
parent | 3a50311c17e4de1998a01bbe939d02fd418fc13a (diff) | |
parent | 6b7697ed486fc3a8e5e7a72344437e66bd4ae507 (diff) |
Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip.c | 6 | ||||
-rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 1 | ||||
-rw-r--r-- | res/res_pjsip_session.c | 24 |
3 files changed, 27 insertions, 4 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index 962c4be4f..e4bcb7038 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> @@ -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> diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index c8ff42708..02562e782 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/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"); diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c index de073d304..5f42dab9f 100644 --- a/res/res_pjsip_session.c +++ b/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, |