Formatting changes only
--Denna och nedanstående rader kommer inte med i loggmeddelandet-- M channels/chan_sip.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
parent
bd30e7abc4
commit
260648043b
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@ -19921,10 +19921,11 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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struct ast_party_connected_line connected;
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struct ast_set_party_connected_line update_connected;
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if (reinvite)
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if (reinvite) {
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ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
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else
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} else {
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ast_debug(4, "SIP response %d to standard invite\n", resp);
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}
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if (p->alreadygone) { /* This call is already gone */
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ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
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@ -19938,8 +19939,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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/* RFC3261 says we must treat every 1xx response (but not 100)
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that we don't recognize as if it was 183.
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*/
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if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183)
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if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183) {
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resp = 183;
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}
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/* For INVITE, treat all 2XX responses as we would a 200 response */
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if ((resp >= 200) && (resp < 300)) {
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@ -19947,16 +19949,19 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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}
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/* Any response between 100 and 199 is PROCEEDING */
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if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
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if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING) {
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p->invitestate = INV_PROCEEDING;
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}
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/* Final response, not 200 ? */
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if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
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if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA )) {
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p->invitestate = INV_COMPLETED;
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}
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/* Final response, clear out pending invite */
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if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
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if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
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p->pendinginvite = 0;
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}
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/* If this is a response to our initial INVITE, we need to set what we can use
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* for this peer.
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@ -19968,15 +19973,17 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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switch (resp) {
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case 100: /* Trying */
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case 101: /* Dialog establishment */
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if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
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if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
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ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
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}
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check_pendings(p);
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break;
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case 180: /* 180 Ringing */
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case 182: /* 182 Queued */
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if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
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if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
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ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
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}
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if (!req->ignore && p->owner) {
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if (get_rpid(p, req)) {
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/* Queue a connected line update */
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@ -20005,8 +20012,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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}
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}
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if (find_sdp(req)) {
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if (p->invitestate != INV_CANCELLED)
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if (p->invitestate != INV_CANCELLED) {
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p->invitestate = INV_EARLY_MEDIA;
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}
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res = process_sdp(p, req, SDP_T38_NONE);
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if (!req->ignore && p->owner) {
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/* Queue a progress frame only if we have SDP in 180 or 182 */
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@ -20037,8 +20045,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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break;
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case 183: /* Session progress */
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if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
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if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
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ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
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}
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if (!req->ignore && p->owner) {
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if (get_rpid(p, req)) {
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/* Queue a connected line update */
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@ -20063,8 +20072,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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sip_handle_cc(p, req, AST_CC_CCNR);
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}
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if (find_sdp(req)) {
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if (p->invitestate != INV_CANCELLED)
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if (p->invitestate != INV_CANCELLED) {
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p->invitestate = INV_EARLY_MEDIA;
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}
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res = process_sdp(p, req, SDP_T38_NONE);
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if (!req->ignore && p->owner) {
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/* Queue a progress frame */
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@ -20084,8 +20094,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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break;
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case 200: /* 200 OK on invite - someone's answering our call */
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if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
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if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
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ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
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}
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p->authtries = 0;
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if (find_sdp(req)) {
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if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
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@ -20136,14 +20147,16 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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update_call_counter(p, DEC_CALL_RINGING);
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parse_ok_contact(p, req);
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/* Save Record-Route for any later requests we make on this dialogue */
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if (!reinvite)
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if (!reinvite) {
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build_route(p, req, 1);
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}
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if(set_address_from_contact(p)) {
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/* Bad contact - we don't know how to reach this device */
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/* We need to ACK, but then send a bye */
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if (!p->route && !req->ignore)
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if (!p->route && !req->ignore) {
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ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
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}
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}
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}
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@ -20151,10 +20164,11 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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if (!req->ignore && p->owner) {
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if (!reinvite) {
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ast_queue_control(p->owner, AST_CONTROL_ANSWER);
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if (sip_cfg.callevents)
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if (sip_cfg.callevents) {
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manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
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"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
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p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
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}
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} else { /* RE-invite */
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if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
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ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
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@ -20166,8 +20180,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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/* It's possible we're getting an 200 OK after we've tried to disconnect
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by sending CANCEL */
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/* First send ACK, then send bye */
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if (!req->ignore)
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if (!req->ignore) {
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ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
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}
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}
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/* Check for Session-Timers related headers */
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@ -20213,20 +20228,23 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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case 401: /* Www auth */
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/* First we ACK */
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->options)
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if (p->options) {
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p->options->auth_type = resp;
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}
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/* Then we AUTH */
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ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
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if (!req->ignore) {
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if (p->authtries < MAX_AUTHTRIES)
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if (p->authtries < MAX_AUTHTRIES) {
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p->invitestate = INV_CALLING;
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}
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if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
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ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", sip_get_header(&p->initreq, "From"));
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pvt_set_needdestroy(p, "failed to authenticate on INVITE");
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sip_alreadygone(p);
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if (p->owner)
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if (p->owner) {
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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}
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}
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}
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break;
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@ -20257,8 +20275,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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/* Could be REFER caused INVITE with replaces */
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ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner)
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if (p->owner) {
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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}
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sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
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break;
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@ -20272,8 +20291,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
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ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
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if (p->owner)
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if (p->owner) {
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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}
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break;
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@ -20300,18 +20320,21 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
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change_t38_state(p, T38_REJECTED);
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/* Try to reset RTP timers */
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/* XXX Why is this commented away??? */
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//ast_rtp_set_rtptimers_onhold(p->rtp);
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/* Trigger a reinvite back to audio */
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transmit_reinvite_with_sdp(p, FALSE, FALSE);
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} else {
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/* We can't set up this call, so give up */
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if (p->owner && !req->ignore)
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if (p->owner && !req->ignore) {
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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}
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pvt_set_needdestroy(p, "received 488 response");
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/* If there's no dialog to end, then mark p as already gone */
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if (!reinvite)
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if (!reinvite) {
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sip_alreadygone(p);
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}
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}
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break;
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case 491: /* Pending */
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@ -20342,12 +20365,14 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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case 405: /* Not allowed */
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case 501: /* Not implemented */
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xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner)
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if (p->owner) {
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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}
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break;
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}
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if (xmitres == XMIT_ERROR)
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if (xmitres == XMIT_ERROR) {
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ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
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}
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}
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/* \brief Handle SIP response in NOTIFY transaction
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