]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
automerge commit
authorAutomerge Script <automerge@asterisk.org>
Fri, 29 Jun 2007 17:17:36 +0000 (17:17 +0000)
committerAutomerge Script <automerge@asterisk.org>
Fri, 29 Jun 2007 17:17:36 +0000 (17:17 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@72662 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_iax2.c

index 70164b9e72a6529a687874e9e183b2a5dd51b6ae..c8007acbab1eb2ead7714edada9381f7cae59854 100644 (file)
@@ -267,7 +267,10 @@ enum {
        IAX_FORCEJITTERBUF =    (1 << 20),      /*!< Force jitterbuffer, even when bridged to a channel that can take jitter */ 
        IAX_RTIGNOREREGEXPIRE = (1 << 21),      /*!< When using realtime, ignore registration expiration */
        IAX_TRUNKTIMESTAMPS =   (1 << 22),      /*!< Send trunk timestamps */
-       IAX_MAXAUTHREQ =        (1 << 23)       /*!< Maximum outstanding AUTHREQ restriction is in place */
+       IAX_MAXAUTHREQ =        (1 << 23),      /*!< Maximum outstanding AUTHREQ restriction is in place */
+       IAX_DELAYPBXSTART =     (1 << 25),      /*!< Don't start a PBX on the channel until the peer sends us a
+                                                    response, so that we've achieved a three-way handshake with
+                                                    them before sending voice or anything else*/
 } iax2_flags;
 
 static int global_rtautoclear = 120;
@@ -3440,7 +3443,7 @@ static int iax2_getpeertrunk(struct sockaddr_in sin)
 }
 
 /*--- ast_iax2_new: Create new call, interface with the PBX core */
-static struct ast_channel *ast_iax2_new(int callno, int state, int capability)
+static struct ast_channel *ast_iax2_new(int callno, int state, int capability, unsigned int delaypbx)
 {
        struct ast_channel *tmp;
        struct chan_iax2_pvt *i;
@@ -3488,8 +3491,10 @@ static struct ast_channel *ast_iax2_new(int callno, int state, int capability)
 
                for (v = i->vars ; v ; v = v->next)
                        pbx_builtin_setvar_helper(tmp, v->name, v->value);
-               
-               if (state != AST_STATE_DOWN) {
+
+               if (delaypbx) {
+                       ast_set_flag(i, IAX_DELAYPBXSTART);
+               } else if (state != AST_STATE_DOWN) {
                        if (ast_pbx_start(tmp)) {
                                ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
                                ast_hangup(tmp);
@@ -6815,6 +6820,25 @@ static int socket_read(int *id, int fd, short events, void *cbdata)
                                f.data = empty;
                        memset(&ies, 0, sizeof(ies));
                }
+
+               /* when we receive the first full frame for a new incoming channel,
+                  it is safe to start the PBX on the channel because we have now
+                  completed a 3-way handshake with the peer */
+               if ((f.frametype == AST_FRAME_VOICE) ||
+                   (f.frametype == AST_FRAME_VIDEO) ||
+                   (f.frametype == AST_FRAME_IAX)) {
+                       if (ast_test_flag(iaxs[fr->callno], IAX_DELAYPBXSTART)) {
+                               ast_clear_flag(iaxs[fr->callno], IAX_DELAYPBXSTART);
+                               if (ast_pbx_start(iaxs[fr->callno]->owner)) {
+                                       ast_log(LOG_WARNING, "Unable to start PBX on %s\n", iaxs[fr->callno]->owner->name);
+                                       ast_hangup(iaxs[fr->callno]->owner);
+                                       iaxs[fr->callno]->owner = NULL;
+                                       ast_mutex_unlock(&iaxsl[fr->callno]);
+                                       return 1;
+                               }
+                       }
+               }
+
                if (f.frametype == AST_FRAME_VOICE) {
                        if (f.subclass != iaxs[fr->callno]->voiceformat) {
                                        iaxs[fr->callno]->voiceformat = f.subclass;
@@ -7077,7 +7101,9 @@ retryowner:
                                                                                                VERBOSE_PREFIX_4,
                                                                                                using_prefs);
                                                                
-                                                               if(!(c = ast_iax2_new(fr->callno, AST_STATE_RING, format)))
+                                                               /* create an Asterisk channel for this call, but don't start
+                                                                  a PBX on it until we have received a full frame from the peer */
+                                                               if (!(c = ast_iax2_new(fr->callno, AST_STATE_RING, format, 1)))
                                                                        iax2_destroy_nolock(fr->callno);
                                                        } else {
                                                                ast_set_flag(&iaxs[fr->callno]->state, IAX_STATE_TBD);
@@ -7486,7 +7512,7 @@ retryowner2:
                                                                                        using_prefs);
 
                                                        ast_set_flag(&iaxs[fr->callno]->state, IAX_STATE_STARTED);
-                                                       if(!(c = ast_iax2_new(fr->callno, AST_STATE_RING, format)))
+                                                       if(!(c = ast_iax2_new(fr->callno, AST_STATE_RING, format, 0)))
                                                                iax2_destroy_nolock(fr->callno);
                                                } else {
                                                        ast_set_flag(&iaxs[fr->callno]->state, IAX_STATE_TBD);
@@ -7514,7 +7540,7 @@ retryowner2:
                                                        ast_verbose(VERBOSE_PREFIX_3 "Accepting DIAL from %s, formats = 0x%x\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), iaxs[fr->callno]->peerformat);
                                                ast_set_flag(&iaxs[fr->callno]->state, IAX_STATE_STARTED);
                                                send_command(iaxs[fr->callno], AST_FRAME_CONTROL, AST_CONTROL_PROGRESS, 0, NULL, 0, -1);
-                                               if(!(c = ast_iax2_new(fr->callno, AST_STATE_RING, iaxs[fr->callno]->peerformat)))
+                                               if(!(c = ast_iax2_new(fr->callno, AST_STATE_RING, iaxs[fr->callno]->peerformat, 0)))
                                                        iax2_destroy_nolock(fr->callno);
                                        }
                                }
@@ -8052,7 +8078,7 @@ static struct ast_channel *iax2_request(const char *type, int format, void *data
        if (cai.found)
                ast_copy_string(iaxs[callno]->host, pds.peer, sizeof(iaxs[callno]->host));
 
-       c = ast_iax2_new(callno, AST_STATE_DOWN, cai.capability);
+       c = ast_iax2_new(callno, AST_STATE_DOWN, cai.capability, 0);
 
        ast_mutex_unlock(&iaxsl[callno]);