automerge commit

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@72662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Automerge Script 18 years ago
parent 38b6686e31
commit 09088bc7cc

@ -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]);

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