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@ -11791,6 +11791,18 @@ static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_requ
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return 1;
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}
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/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
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static void stop_data_flows(struct sip_pvt *p)
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{
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/* Immediately stop RTP, VRTP and UDPTL as applicable */
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if (p->rtp)
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ast_rtp_stop(p->rtp);
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if (p->vrtp)
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ast_rtp_stop(p->vrtp);
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if (p->udptl)
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ast_udptl_stop(p->udptl);
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}
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/*! \brief Handle SIP response in dialogue */
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/* XXX only called by handle_request */
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static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno)
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@ -11980,18 +11992,9 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
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if ((option_verbose > 2) && (resp != 487))
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ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
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ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
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if (p->rtp) {
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/* Immediately stop RTP */
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ast_rtp_stop(p->rtp);
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}
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if (p->vrtp) {
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/* Immediately stop VRTP */
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ast_rtp_stop(p->vrtp);
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}
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if (p->udptl) {
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/* Immediately stop UDPTL */
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ast_udptl_stop(p->udptl);
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}
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stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
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/* XXX Locking issues?? XXX */
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switch(resp) {
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case 300: /* Multiple Choices */
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@ -13721,18 +13724,8 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
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ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n");
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return 0;
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}
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if (p->rtp) {
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/* Immediately stop RTP */
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ast_rtp_stop(p->rtp);
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}
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if (p->vrtp) {
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/* Immediately stop VRTP */
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ast_rtp_stop(p->vrtp);
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}
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if (p->udptl) {
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/* Immediately stop UDPTL */
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ast_udptl_stop(p->udptl);
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}
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stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
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if (p->owner)
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ast_queue_hangup(p->owner);
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else
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@ -13753,7 +13746,6 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
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struct ast_channel *c=NULL;
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int res;
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struct ast_channel *bridged_to;
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char *audioqos = NULL, *videoqos = NULL;
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if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE))
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transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
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@ -13762,35 +13754,27 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
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check_via(p, req);
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ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
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if (p->rtp)
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audioqos = ast_rtp_get_quality(p->rtp);
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if (p->vrtp)
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videoqos = ast_rtp_get_quality(p->vrtp);
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/* Get RTCP quality before end of call */
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if (recordhistory) {
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if (p->rtp)
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append_history(p, "RTCPaudio", "Quality:%s", audioqos);
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if (p->vrtp)
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append_history(p, "RTCPvideo", "Quality:%s", videoqos);
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}
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if (recordhistory || p->owner) {
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char *audioqos, *videoqos;
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if (p->rtp) {
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audioqos = ast_rtp_get_quality(p->rtp);
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if (recordhistory)
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append_history(p, "RTCPaudio", "Quality:%s", audioqos);
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if (p->owner)
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pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
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/* Immediately stop RTP */
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ast_rtp_stop(p->rtp);
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}
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if (p->vrtp) {
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videoqos = ast_rtp_get_quality(p->vrtp);
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if (recordhistory)
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append_history(p, "RTCPvideo", "Quality:%s", videoqos);
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if (p->owner)
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pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
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/* Immediately stop VRTP */
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ast_rtp_stop(p->vrtp);
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}
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if (p->udptl) {
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/* Immediately stop UDPTL */
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ast_udptl_stop(p->udptl);
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}
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stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
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if (!ast_strlen_zero(get_header(req, "Also"))) {
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ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
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ast_inet_ntoa(p->recv.sin_addr));
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