automerge commit

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@12538 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6a62ea5c614f79a5ee2977d195e22b619f03da85..844b7b2bba00f6ec9f372d62474ee1b1814c6f2e 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2485,13 +2485,15 @@ static int sip_hangup(struct ast_channel *ast)
        if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
                if (needcancel) {       /* Outgoing call, not up */
                        if (ast_test_flag(p, SIP_OUTGOING)) {
+                               /* stop retransmitting an INVITE that has not received a response */
+                               __sip_pretend_ack(p);
+
+                               /* Send a new request: CANCEL */
                                transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
                                /* Actually don't destroy us yet, wait for the 487 on our original 
                                   INVITE, but do set an autodestruct just in case we never get it. */
                                ast_clear_flag(&locflags, SIP_NEEDDESTROY);
-                               sip_scheddestroy(p, 15000);
-                               /* stop retransmitting an INVITE that has not received a response */
-                               __sip_pretend_ack(p);
+                               sip_scheddestroy(p, 32000);
                                if ( p->initid != -1 ) {
                                        /* channel still up - reverse dec of inUse counter
                                           only if the channel is not auto-congested */
@@ -2521,12 +2523,34 @@ static int sip_hangup(struct ast_channel *ast)
        return 0;
 }
 
+/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
+static void try_suggested_sip_codec(struct sip_pvt *p)
+{
+       int fmt;
+       char *codec;
+
+       codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+       if (!codec) 
+               return;
+
+       fmt = ast_getformatbyname(codec);
+       if (fmt) {
+               ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
+               if (p->jointcapability & fmt) {
+                       p->jointcapability &= fmt;
+                       p->capability &= fmt;
+               } else
+                       ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+       } else
+               ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
+       return; 
+}
+
 /*! \brief  sip_answer: Answer SIP call , send 200 OK on Invite 
  * Part of PBX interface */
 static int sip_answer(struct ast_channel *ast)
 {
-       int res = 0,fmt;
-       char *codec;
+       int res = 0;
        struct sip_pvt *p = ast->tech_pvt;
 
        ast_mutex_lock(&p->lock);
@@ -2534,19 +2558,7 @@ static int sip_answer(struct ast_channel *ast)
 #ifdef OSP_SUPPORT     
                time(&p->ospstart);
 #endif
-       
-               codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
-               if (codec) {
-                       fmt=ast_getformatbyname(codec);
-                       if (fmt) {
-                               ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
-                               if (p->jointcapability & fmt) {
-                                       p->jointcapability &= fmt;
-                                       p->capability &= fmt;
-                               } else
-                                       ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
-                       } else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
-               }
+               try_suggested_sip_codec(p);     
 
                ast_setstate(ast, AST_STATE_UP);
                if (option_debug)
@@ -4580,6 +4592,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
        respprep(&resp, p, msg, req);
        if (p->rtp) {
                ast_rtp_offered_from_local(p->rtp, 0);
+               try_suggested_sip_codec(p);     
                add_sdp(&resp, p);
        } else {
                ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
@@ -9329,6 +9342,8 @@ static char *function_sippeer(struct ast_channel *chan, char *cmd, char *data, c
                snprintf(buf, len, "%d", peer->call_limit);
        } else  if (!strcasecmp(colname, "curcalls")) {
                snprintf(buf, len, "%d", peer->inUse);
+       } else  if (!strcasecmp(colname, "accountcode")) {
+               ast_copy_string(buf, peer->accountcode, len);
        } else  if (!strcasecmp(colname, "useragent")) {
                ast_copy_string(buf, peer->useragent, len);
        } else  if (!strcasecmp(colname, "mailbox")) {
@@ -9388,6 +9403,7 @@ struct ast_custom_function sippeer_function = {
        "- curcalls              Current amount of calls \n"
        "                        Only available if call-limit is set\n"
        "- language              Default language for peer\n"
+       "- accountcode           Account code for this peer\n"
        "- useragent             Current user agent id for peer\n"
        "- codec[x]              Preferred codec index number 'x' (beginning with zero).\n"
        "\n"
@@ -9549,12 +9565,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                break;
        case 183:       /* Session progress */
                sip_cancel_destroy(p);
+               /* Ignore 183 Session progress without SDP */
                if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
                        process_sdp(p, req);
-               }
-               if (!ignore && p->owner) {
-                       /* Queue a progress frame */
-                       ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+                       if (!ignore && p->owner) {
+                               /* Queue a progress frame */
+                               ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+                       }
                }
                break;
        case 200:       /* 200 OK on invite - someone's answering our call */