move some dialog-only flags to proper variables, namely
SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, SIP_PAGE2_NOTEXT, SIP_PAGE2_OUTGOING_CALL These are seldom used so the diff is relatively small. Note that 'OUTGOING_CALL' is dangerously similar to another dialog flag, 'SIP_OUTGOING', so the description will need to clarify the different meaning of the two. Also note that the description of NOTEXT is a bit unclear - does it mean we don't support it, or 'not requested or not supported' ? On passing fix a comment referring to video instead of text. Finally, mark with XXX a possibly misleading debugging message. (maybe the latter is worth backporting). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77739 f38db490-d61c-443f-a65b-d21fe96a405b
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630c2fe170
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1 changed files with 19 additions and 17 deletions
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@ -775,7 +775,6 @@ struct sip_auth {
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they have a common layout so it is easy to copy them.
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*/
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#define SIP_OUTGOING (1 << 0) /*!< D: Direction of the last transaction in this dialog */
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#define SIP_NOVIDEO (1 << 1) /*!< D: Didn't get video in invite, don't offer */
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#define SIP_RINGING (1 << 2) /*!< D: Have sent 180 ringing */
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#define SIP_PROGRESS_SENT (1 << 3) /*!< D: Have sent 183 message progress */
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#define SIP_NEEDREINVITE (1 << 4) /*!< D: Do we need to send another reinvite? */
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@ -784,7 +783,6 @@ struct sip_auth {
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#define SIP_CALL_LIMIT (1 << 7) /*!< D: Call limit enforced for this call */
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#define SIP_INC_COUNT (1 << 8) /*!< D: Did this dialog increment the counter of in-use calls? */
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#define SIP_INC_RINGING (1 << 9) /*!< D: Did this connection increment the counter of in-use calls? */
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#define SIP_DIALOG_ANSWEREDELSEWHERE (1 << 10) /*!< D: This call is cancelled due to answer on another channel */
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#define SIP_DEFER_BYE_ON_TRANSFER (1 << 11) /*!< D: Do not hangup at first ast_hangup */
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#define SIP_PROMISCREDIR (1 << 12) /*!< DP: Promiscuous redirection */
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@ -855,9 +853,7 @@ struct sip_auth {
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#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for buggy RFC2833 implementations */
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#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO MWI fix */
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#define SIP_PAGE2_NOTEXT (1 << 27) /*!< GDP: Text not supported */
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#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< GDP: Global text enable */
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#define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< D: Is this an outgoing call? */
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#define SIP_PAGE2_FLAGS_TO_COPY \
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(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
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@ -1049,6 +1045,10 @@ struct sip_pvt {
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char do_history; /*!< Set if we want to record history */
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char alreadygone; /*!< already destroyed by our peer */
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char needdestroy; /*!< need to be destroyed by the monitor thread */
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char outgoing_call; /*!< this is an outgoing call */
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char answered_elsewhere; /*!< This call is cancelled due to answer on another channel */
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char novideo; /*!< Didn't get video in invite, don't offer */
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char notext; /*!< Text not supported (?) */
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int timer_t1; /*!< SIP timer T1, ms rtt */
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unsigned int sipoptions; /*!< Supported SIP options on the other end */
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@ -3596,7 +3596,7 @@ static int update_call_counter(struct sip_pvt *fup, int event)
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{
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char name[256];
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int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
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int outgoing = ast_test_flag(&fup->flags[1], SIP_PAGE2_OUTGOING_CALL);
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int outgoing = fup->outgoing_call;
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struct sip_user *u = NULL;
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struct sip_peer *p = NULL;
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@ -3883,7 +3883,7 @@ static int sip_hangup(struct ast_channel *ast)
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if (option_debug)
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ast_log(LOG_DEBUG, "This call was answered elsewhere");
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append_history(p, "Cancel", "Call answered elsewhere");
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ast_set_flag(&p->flags[0], SIP_DIALOG_ANSWEREDELSEWHERE);
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p->answered_elsewhere = TRUE;
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}
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if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
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@ -4347,7 +4347,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
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ast_moh_stop(ast);
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break;
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case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
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if (p->vrtp && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
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if (p->vrtp && !p->novideo) {
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transmit_info_with_vidupdate(p);
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/* ast_rtcp_send_h261fur(p->vrtp); */
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} else
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@ -5451,11 +5451,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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return -1;
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}
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vhp = hp; /* Copy to video address as default too */
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thp = hp; /* Copy to video address as default too */
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thp = hp; /* Copy to text address as default too */
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iterator = req->sdp_start;
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ast_set_flag(&p->flags[0], SIP_NOVIDEO);
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ast_set_flag(&p->flags[1], SIP_PAGE2_NOTEXT);
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/* default: novideo and notext set */
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p->novideo = TRUE;
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p->notext = TRUE;
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if (p->vrtp)
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ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
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@ -5490,7 +5491,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
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(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
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video = TRUE;
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ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
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p->novideo = FALSE;
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numberofmediastreams++;
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vportno = x;
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/* Scan through the RTP payload types specified in a "m=" line: */
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@ -5506,7 +5507,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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} else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
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(sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1)) {
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text = TRUE;
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ast_clear_flag(&p->flags[1], SIP_PAGE2_NOTEXT);
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p->notext = FALSE;
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numberofmediastreams++;
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tportno = x;
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/* Scan through the RTP payload types specified in a "m=" line: */
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@ -6960,8 +6961,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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capability = p->jointcapability;
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/* XXX note, Video and Text are negated - 'true' means 'no' */
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ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
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ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False");
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p->novideo ? "True" : "False", p->notext ? "True" : "False");
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ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
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#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
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@ -6972,7 +6974,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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#endif
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/* Check if we need video in this call */
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if ((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
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if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
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if (p->vrtp) {
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needvideo = TRUE;
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ast_debug(2, "This call needs video offers!\n");
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@ -6999,7 +7001,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
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}
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/* Check if we need text in this call */
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if((capability & AST_FORMAT_TEXT_MASK) && !ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT)) {
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if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
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if (sipdebug_text)
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ast_verbose("We think we can do text\n");
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if (p->trtp) {
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@ -8334,7 +8336,7 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
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p->invitestate = INV_CONFIRMED;
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reqprep(&resp, p, sipmethod, seqno, newbranch);
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if (sipmethod == SIP_CANCEL && ast_test_flag(&p->flags[0], SIP_DIALOG_ANSWEREDELSEWHERE))
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if (sipmethod == SIP_CANCEL && p->answered_elsewhere)
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add_header(&resp, "Reason:", "SIP;cause=200;text=\"Call completed elsewhere\"");
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add_header_contentLength(&resp, 0);
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@ -16592,7 +16594,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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return NULL;
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}
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ast_set_flag(&p->flags[1], SIP_PAGE2_OUTGOING_CALL);
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p->outgoing_call = TRUE;
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if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
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sip_destroy(p);
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