aboutsummaryrefslogtreecommitdiffstats
path: root/channels
diff options
context:
space:
mode:
authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-09-05 23:48:48 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-09-05 23:48:48 +0000
commit2993593622bec2056db9560074d9aff5609fa59d (patch)
tree5591386be584106884a4ed675d417a70c15fe946 /channels
parent41ca1c9fb93018c5c83aba6816f499f841b41054 (diff)
Merged revisions 41768,41827,41830,41880,41882,41989,42014,42054 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r41768 | file | 2006-09-01 18:49:07 -0400 (Fri, 01 Sep 2006) | 2 lines Only wipe the redirected audio & video IP/port if it's specified, and trigger a reinvite. ........ r41827 | bweschke | 2006-09-03 10:16:08 -0400 (Sun, 03 Sep 2006) | 3 lines Setting a retry of 0 is generally not a good idea and shouldn't be allowed. (#7574 - reported by regin) ........ r41830 | bweschke | 2006-09-03 10:50:59 -0400 (Sun, 03 Sep 2006) | 3 lines Let's NOT spy on Zap/psuedo channels, mmmmmmmmk? ........ r41880 | bweschke | 2006-09-03 13:13:38 -0400 (Sun, 03 Sep 2006) | 3 lines Don't keep trying the same member in certain strategies when members of the queue are unavailable (#7278 - diLLec reported and patched) - This should have been patched here first and then merged into /trunk. My bad! ........ r41882 | bweschke | 2006-09-03 13:38:22 -0400 (Sun, 03 Sep 2006) | 3 lines Make sure the forwarded channel inherits variables appropriately when we receive a call forward in the queue. (#7867 - raarts reported and patched) ........ r41989 | oej | 2006-09-04 11:46:07 -0400 (Mon, 04 Sep 2006) | 2 lines Don't kill the pvt before we have sent ACK on CANCEL (needs more testing before making a release) ........ r42014 | qwell | 2006-09-05 12:27:46 -0400 (Tue, 05 Sep 2006) | 4 lines Small typo in zapata.conf.sample Reported by ppyy in 7881 ........ r42054 | file | 2006-09-05 16:02:48 -0400 (Tue, 05 Sep 2006) | 2 lines Merge in last round of spy fixes. This should hopefully eliminate all the issues people have been seeing by distinctly separating what each component (core/spy) is responsible for. Core is responsible for adding a spy to a channel, feeding frames to the spy, removing the spy from a channel, and telling the spy to stop. Spy is responsible for reading frames in, and cleaning up after itself. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2-netsec@42081 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c55
1 files changed, 38 insertions, 17 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5259a459f..d217ba81c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2447,7 +2447,7 @@ static int sip_hangup(struct ast_channel *ast)
{
struct sip_pvt *p = ast->tech_pvt;
int needcancel = 0;
- struct ast_flags locflags = {0};
+ int needdestroy = 0;
if (!p) {
ast_log(LOG_DEBUG, "Asked to hangup channel not connected\n");
@@ -2481,7 +2481,6 @@ static int sip_hangup(struct ast_channel *ast)
#endif
/* Disconnect */
- p = ast->tech_pvt;
if (p->vad) {
ast_dsp_free(p->vad);
}
@@ -2493,7 +2492,16 @@ static int sip_hangup(struct ast_channel *ast)
ast_mutex_unlock(&usecnt_lock);
ast_update_use_count();
- ast_set_flag(&locflags, SIP_NEEDDESTROY);
+ /* Do not destroy this pvt until we have timeout or
+ get an answer to the BYE or INVITE/CANCEL
+ If we get no answer during retransmit period, drop the call anyway.
+ (Sorry, mother-in-law, you can't deny a hangup by sending
+ 603 declined to BYE...)
+ */
+ if (ast_test_flag(p, SIP_ALREADYGONE))
+ needdestroy = 1; /* Set destroy flag at end of this function */
+ else
+ sip_scheddestroy(p, 32000);
/* Start the process if it's not already started */
if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
@@ -2506,13 +2514,12 @@ static int sip_hangup(struct ast_channel *ast)
it pending */
if (!ast_test_flag(p, SIP_CAN_BYE)) {
ast_set_flag(p, SIP_PENDINGBYE);
+ /* Do we need a timer here if we don't hear from them at all? */
} else {
/* 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, 32000);
}
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
@@ -2538,7 +2545,8 @@ static int sip_hangup(struct ast_channel *ast)
}
}
}
- ast_copy_flags(p, (&locflags), SIP_NEEDDESTROY);
+ if (needdestroy)
+ ast_set_flag(p, SIP_NEEDDESTROY);
ast_mutex_unlock(&p->lock);
return 0;
}
@@ -10085,6 +10093,9 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
handle_response_invite(p, resp, rest, req, ignore, seqno);
} else if (sipmethod == SIP_REGISTER) {
res = handle_response_register(p, resp, rest, req, ignore, seqno);
+ } else if (sipmethod == SIP_BYE) {
+ /* Ok, we're ready to go */
+ ast_set_flag(p, SIP_NEEDDESTROY);
}
break;
case 401: /* Not www-authorized on SIP method */
@@ -10236,9 +10247,12 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
handle_response_invite(p, resp, rest, req, ignore, seqno);
} else if (sipmethod == SIP_CANCEL) {
ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n");
- } else if (sipmethod == SIP_MESSAGE)
+ } else if (sipmethod == SIP_MESSAGE)
/* We successfully transmitted a message */
ast_set_flag(p, SIP_NEEDDESTROY);
+ else if (sipmethod == SIP_BYE)
+ /* Ok, we're ready to go */
+ ast_set_flag(p, SIP_NEEDDESTROY);
break;
case 401: /* www-auth */
case 407:
@@ -10924,10 +10938,15 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req, int de
if (p->owner)
ast_queue_hangup(p->owner);
}
- } else if (p->owner)
+ } else if (p->owner) {
ast_queue_hangup(p->owner);
- else
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Received bye, issuing owner hangup\n.");
+ } else {
ast_set_flag(p, SIP_NEEDDESTROY);
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Received bye, no owner, selfdestruct soon.\n.");
+ }
transmit_response(p, "200 OK", req);
return 1;
@@ -13057,24 +13076,26 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
if (!p)
return -1;
ast_mutex_lock(&p->lock);
- if (rtp) {
+ if (rtp) {
changed |= ast_rtp_get_peer(rtp, &p->redirip);
#ifdef SIP_MIDCOM
if (m_cb)
- m_cb->ast_rtp_get_their_nat_audio_hook(rtp, p->r);
+ m_cb->ast_rtp_get_their_nat_audio_hook(rtp, p->r);
#endif
- }
- else
+ } else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
memset(&p->redirip, 0, sizeof(p->redirip));
- if (vrtp) {
+ changed = 1;
+ }
+ if (vrtp) {
changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
#ifdef SIP_MIDCOM
if (m_cb)
- m_cb->ast_rtp_get_their_nat_video_hook(vrtp, p->r);
+ m_cb->ast_rtp_get_their_nat_video_hook(vrtp, p->r);
#endif
- }
- else
+ } else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
memset(&p->vredirip, 0, sizeof(p->vredirip));
+ changed = 1;
+ }
if (codecs && (p->redircodecs != codecs)) {
p->redircodecs = codecs;
changed = 1;