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small formatting cleanup
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -4500,6 +4500,8 @@ static int find_sdp(struct sip_request *req)
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/*! \brief Process SIP SDP offer, select formats and activate RTP channels
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If offer is rejected, we will not change any properties of the call
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Return 0 on success, a negative value on errors.
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Must be called after find_sdp().
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*/
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static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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{
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@@ -8712,6 +8714,16 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
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if (ast_strlen_zero(of))
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return AUTH_SUCCESSFUL;
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if (1) {
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const char *hdr = (sipmethod == SIP_REGISTER || sipmethod == SIP_SUBSCRIBE) ?
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"Authorization" : "Proxy-Authorization";
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if ( (hdr = get_header(req, hdr)) && (hdr = strstr(hdr, "username=\"")) ) {
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ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
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of = from;
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of = strsep(&of, "\"");
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}
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}
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if (!authpeer) /* If we are looking for a peer, don't check the user objects (or realtime) */
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user = find_user(of, 1);
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@@ -11285,10 +11297,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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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) &&
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(resp < 200) &&
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(resp != 180) &&
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(resp != 183))
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if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
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resp = 183;
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switch (resp) {
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@@ -11297,6 +11306,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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sip_cancel_destroy(p);
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check_pendings(p);
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break;
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case 180: /* 180 Ringing */
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if (!ast_test_flag(req, SIP_PKT_IGNORE))
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sip_cancel_destroy(p);
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@@ -11316,6 +11326,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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ast_set_flag(&p->flags[0], SIP_CAN_BYE);
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check_pendings(p);
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break;
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case 183: /* Session progress */
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if (!ast_test_flag(req, SIP_PKT_IGNORE))
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sip_cancel_destroy(p);
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@@ -11330,6 +11341,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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ast_set_flag(&p->flags[0], SIP_CAN_BYE);
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check_pendings(p);
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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 (!ast_test_flag(req, SIP_PKT_IGNORE))
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sip_cancel_destroy(p);
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@@ -11428,6 +11440,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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ast_set_flag(&p->flags[0], SIP_CAN_BYE);
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check_pendings(p);
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break;
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case 407: /* Proxy authentication */
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case 401: /* Www auth */
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/* First we ACK */
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@@ -11449,6 +11462,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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}
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}
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break;
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case 403: /* Forbidden */
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/* First we ACK */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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@@ -11458,22 +11472,26 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
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ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
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break;
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case 404: /* Not found */
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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
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break;
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case 481: /* Call leg does not exist */
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/* Could be REFER or INVITE */
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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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transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
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break;
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case 491: /* Pending */
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/* we have to wait a while, then retransmit */
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/* Transmission is rescheduled, so everything should be taken care of.
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We should support the retry-after at some point */
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break;
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case 501: /* Not implemented */
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if (p->owner)
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ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
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