[svn-commits] oej: branch oej/codename-pineapple r47342 - in
/team/oej/codename-pineapple/c...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed Nov 8 15:47:56 MST 2006
Author: oej
Date: Wed Nov 8 16:47:55 2006
New Revision: 47342
URL: http://svn.digium.com/view/asterisk?rev=47342&view=rev
Log:
Update
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
team/oej/codename-pineapple/channels/sip3/sip3.h
team/oej/codename-pineapple/channels/sip3/sip3_compose.c
team/oej/codename-pineapple/channels/sip3/sip3_dialog.c
team/oej/codename-pineapple/channels/sip3/sip3_network.c
team/oej/codename-pineapple/channels/sip3/sip3_refer.c
team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
team/oej/codename-pineapple/channels/sip3/sip3funcs.h
Modified: team/oej/codename-pineapple/channels/chan_sip3.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_sip3.c?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Wed Nov 8 16:47:55 2006
@@ -166,7 +166,7 @@
- authuser as a separate config option, please, please
- Split up source code file
- Add astum
- - Add T1 configuration settings
+ - Add T1 timer configuration settings
- Add auto-nat for RFC 1918 networks
- Add type=device for peers
- Add type=service for register= replacement
@@ -185,6 +185,10 @@
- Add File's multithreading code
- Make debugaddr a ha list instead of one address and move it out of sipnet
+ Maybe
+ - add support for Path header
+ the Path is arriving with Register requests, saved in location
+ and used as a Route: header in the outbound request
- ... And much more
*/
@@ -393,6 +397,7 @@
#define SIPLABEL sip3
+/*------- GLOBAL VARIABLES ------------------------------------ */
/*! \brief various expiry times for registrations */
struct expiry_times expiry = {
@@ -412,30 +417,26 @@
/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
when it's doing something critical. */
-
AST_MUTEX_DEFINE_STATIC(monlock);
+/*! \brief Make sure we don't reload twice at the same time */
AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
/*! \brief This is the thread for the monitor which checks for input on the channels
which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
-static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
-static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
+static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
+static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
struct sched_context *sched; /*!< The scheduling context */
struct io_context *io; /*!< The IO context */
/* --- Linked lists of various objects --------*/
-/*! \brief List of concurrent SIP dialogs */
-struct sip_dialog *dialoglist = NULL;
-
-/*! \brief The peer list: Peers and Friends */
-struct sip_device_list devicelist;
-
-struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
+struct sip_dialog *dialoglist = NULL; /*!< List of concurrent SIP dialogs */
+struct sip_device_list devicelist; /*!< The device list */
+struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
/*---------------------------- Forward declarations of functions in chan_sip3.c */
@@ -485,8 +486,8 @@
/*--- Authentication stuff */
static enum check_auth_result check_user_full(struct sip_dialog *p, struct sip_request *req,
- int sipmethod, char *uri, enum xmittype reliable,
- struct sockaddr_in *sin, struct sip_peer **authpeer);
+ int sipmethod, char *uri, enum xmittype reliable,
+ struct sockaddr_in *sin, struct sip_peer **authpeer);
static int check_user(struct sip_dialog *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
/*--- Misc functions */
@@ -561,9 +562,6 @@
static void handle_response_refer(struct sip_dialog *p, int resp, char *rest, struct sip_request *req);
static void handle_response_peerpoke(struct sip_dialog *p, int resp, struct sip_request *req);
static void handle_response(struct sip_dialog *p, int resp, char *rest, struct sip_request *req);
-
-/*----- RTP interface functions */
-static void stop_media_flows(struct sip_dialog *p);
/*------ T38 Support --------- */
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_dialog *pvt, int reinvite); /*!< T38 negotiation helper function */
@@ -1102,7 +1100,7 @@
{
struct sip_dialog *dialog = nothing;
- ast_mutex_lock(&dialog->lock);
+ dialog_lock(dialog, TRUE);
dialog->initid = -1;
if (dialog->owner) {
/* XXX fails on possible deadlock */
@@ -1113,7 +1111,7 @@
ast_channel_unlock(dialog->owner);
}
}
- ast_mutex_unlock(&dialog->lock);
+ dialog_lock(dialog, FALSE);
return 0;
}
@@ -1331,7 +1329,7 @@
if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE))
ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
if (option_debug && sipdebug)
ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->peername);
update_call_counter(p, DEC_CALL_LIMIT);
@@ -1339,7 +1337,7 @@
/* Determine how to disconnect */
if (p->owner != ast) {
ast_log(LOG_WARNING, "Huh? We aren't the owner? Can't hangup call.\n");
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return 0;
}
/* If the call is not UP, we need to send CANCEL instead of BYE */
@@ -1433,7 +1431,7 @@
}
if (needdestroy)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return 0;
}
@@ -1467,7 +1465,7 @@
int res = 0;
struct sip_dialog *dialog = ast->tech_pvt;
- ast_mutex_lock(&dialog->lock);
+ dialog_lock(dialog, TRUE);
if (ast->_state != AST_STATE_UP) {
try_suggested_sip_codec(dialog);
@@ -1483,7 +1481,7 @@
res = transmit_response_with_attachment(WITH_SDP, dialog, "200 OK", &dialog->initreq, XMIT_CRITICAL);
dialogstatechange(dialog, DIALOG_STATE_CONFIRMED);
}
- ast_mutex_unlock(&dialog->lock);
+ dialog_lock(dialog, FALSE);
return res;
}
@@ -1508,7 +1506,7 @@
return 0;
}
if (p) {
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
if (p->rtp) {
/* If channel is not up, activate early media session */
if ((ast->_state != AST_STATE_UP) &&
@@ -1520,12 +1518,12 @@
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->rtp, frame);
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
}
break;
case AST_FRAME_VIDEO:
if (p) {
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
if (p->vrtp) {
/* Activate video early media */
if ((ast->_state != AST_STATE_UP) &&
@@ -1537,7 +1535,7 @@
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->vrtp, frame);
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
}
break;
case AST_FRAME_IMAGE:
@@ -1545,7 +1543,7 @@
break;
case AST_FRAME_MODEM:
if (p) {
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
if (p->udptl) {
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
@@ -1555,7 +1553,7 @@
}
res = ast_udptl_write(p->udptl, frame);
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
}
break;
default:
@@ -1587,7 +1585,7 @@
}
p = newchan->tech_pvt;
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
append_history(p, "Masq", "Old channel: %s\n", oldchan->name);
append_history(p, "Masq (cont)", "...new owner: %s\n", newchan->name);
if (p->owner != oldchan)
@@ -1599,7 +1597,7 @@
if (option_debug > 2)
ast_log(LOG_DEBUG, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return ret;
}
@@ -1610,7 +1608,7 @@
struct sip_dialog *p = ast->tech_pvt;
int res = 0;
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
case SIP_DTMF_INBAND:
res = -1; /* Tell Asterisk to generate inband indications */
@@ -1622,7 +1620,7 @@
default:
break;
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return res;
}
@@ -1634,7 +1632,7 @@
struct sip_dialog *p = ast->tech_pvt;
int res = 0;
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
case SIP_DTMF_INFO:
transmit_info_with_digit(p, digit);
@@ -1647,7 +1645,7 @@
res = -1; /* Tell Asterisk to stop inband indications */
break;
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return res;
}
@@ -1660,12 +1658,12 @@
if (dest == NULL)
dest = "";
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
if (ast->_state == AST_STATE_RING)
res = sip_sipredirect(p, dest);
else
res = transmit_refer(p, dest);
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return res;
}
@@ -1679,7 +1677,7 @@
struct sip_dialog *p = ast->tech_pvt;
int res = 0;
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
switch(condition) {
case AST_CONTROL_RINGING:
if (ast->_state == AST_STATE_RING) {
@@ -1754,7 +1752,7 @@
res = -1;
break;
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return res;
}
@@ -1773,10 +1771,23 @@
int what;
int needvideo = 0;
- dialoglist_unlock();
- /* Don't hold a sip pvt lock while we allocate a channel */
- tmp = ast_channel_alloc(1);
- dialoglist_lock();
+ {
+ const char *my_name; /* pick a good name */
+
+ if (title)
+ my_name = title;
+ else if ( (my_name = strchr(dialog->fromdomain,':')) )
+ my_name++; /* skip ':' */
+ else
+ my_name = dialog->fromdomain;
+
+ dialog_lock(dialog, FALSE);
+ /* Don't hold a sip pvt lock while we allocate a channel */
+
+ tmp = ast_channel_alloc(1, state, dialog->cid_num, dialog->cid_name, "SIP/%s-%08x", my_name, (int)(long) dialog);
+
+ }
+ dialog_lock(dialog, TRUE);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
return NULL;
@@ -1823,18 +1834,6 @@
ast_log(LOG_DEBUG, "This channel can handle video! HOLLYWOOD next!\n");
else
ast_log(LOG_DEBUG, "This channel will not be able to handle video.\n");
- }
-
-
- {
- const char *my_name; /* pick a good name */
- if (title)
- my_name = title;
- else if ( (my_name = strchr(dialog->fromdomain,':')) )
- my_name++; /* skip ':' */
- else
- my_name = dialog->fromdomain;
- ast_string_field_build(tmp, name, "SIP/%s-%08x", my_name, (int)(long) dialog);
}
if (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
@@ -1894,7 +1893,6 @@
pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", dialog->domain);
if (!ast_strlen_zero(dialog->callid))
pbx_builtin_setvar_helper(tmp, "SIPCALLID", dialog->callid);
- ast_setstate(tmp, state);
if (dialog->rtp)
ast_jb_configure(tmp, &global.jbconf);
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
@@ -3557,7 +3555,7 @@
char *ourtag = sip_dialog_ptr->tag;
/* Go ahead and lock it (and its owner) before returning */
- ast_mutex_lock(&sip_dialog_ptr->lock);
+ dialog_lock(sip_dialog_ptr, TRUE);
/* Check if tags match. If not, this is not the call we want
(With a forking SIP proxy, several call legs share the
@@ -3567,7 +3565,7 @@
match = 0;
if (!match) {
- ast_mutex_unlock(&sip_dialog_ptr->lock);
+ dialog_lock(sip_dialog_ptr, FALSE);
continue;
}
@@ -3578,9 +3576,9 @@
/* deadlock avoidance... */
while (sip_dialog_ptr->owner && ast_channel_trylock(sip_dialog_ptr->owner)) {
- ast_mutex_unlock(&sip_dialog_ptr->lock);
+ dialog_lock(sip_dialog_ptr, FALSE);
usleep(1);
- ast_mutex_lock(&sip_dialog_ptr->lock);
+ dialog_lock(sip_dialog_ptr, TRUE);
}
break;
}
@@ -4667,9 +4665,9 @@
} else {
if (option_debug > 1)
ast_log(LOG_DEBUG, "Strange... The other side of the bridge does not have a udptl struct\n");
- ast_mutex_lock(&bridgepvt->lock);
+ dialog_lock(bridgepvt, TRUE);
bridgepvt->t38.state = T38_DISABLED;
- ast_mutex_unlock(&bridgepvt->lock);
+ dialog_lock(bridgepvt, FALSE);
if (option_debug)
ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->tech->type);
p->t38.state = T38_DISABLED;
@@ -5479,7 +5477,7 @@
transmit_response_with_attachment(WITH_SDP, p, "200 OK", req, XMIT_RELIABLE);
/* Do something more clever here */
ast_channel_unlock(c);
- ast_mutex_unlock(&p->refer->refer_call->lock);
+ dialog_lock(p->refer->refer_call, FALSE);
return 1;
}
if (!c) {
@@ -5488,7 +5486,7 @@
transmit_response_reliable(p, "503 Service Unavailable", req);
append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- ast_mutex_unlock(&p->refer->refer_call->lock);
+ dialog_lock(p->refer->refer_call, FALSE);
return 1;
}
append_history(p, "Xfer", "INVITE/Replace received");
@@ -5521,8 +5519,8 @@
/* Unlock clone, but not original (replacecall) */
ast_channel_unlock(c);
- /* Unlock PVT */
- ast_mutex_unlock(&p->refer->refer_call->lock);
+ /* Unlock dialog */
+ dialog_lock(p->refer->refer_call, FALSE);
/* Make sure that the masq does not free our PVT for the old call */
ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
@@ -5563,7 +5561,7 @@
}
ast_channel_unlock(replacecall);
}
- ast_mutex_unlock(&p->refer->refer_call->lock);
+ dialog_lock(p->refer->refer_call, FALSE);
ast_setstate(c, AST_STATE_DOWN);
if (option_debug > 3) {
@@ -5585,7 +5583,7 @@
}
ast_channel_unlock(p->owner); /* Unlock new owner */
- ast_mutex_unlock(&p->lock); /* Unlock SIP structure */
+ dialog_lock(p, FALSE);
/* The call should be down with no ast_channel, so hang it up */
c->tech_pvt = NULL;
@@ -5746,9 +5744,9 @@
if (error) { /* Give up this dialog */
append_history(p, "Xfer", "INVITE/Replace Failed.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
if (p->refer->refer_call) {
- ast_mutex_unlock(&p->refer->refer_call->lock);
+ dialog_lock(p->refer->refer_call, FALSE);
ast_channel_unlock(p->refer->refer_call->owner);
}
return -1;
@@ -5945,10 +5943,10 @@
if (res) {
/* Unlock locks so ast_hangup can do its magic */
- ast_mutex_unlock(&c->lock);
- ast_mutex_unlock(&p->lock);
+ ast_channel_unlock(c);
+ dialog_lock(p, FALSE);
ast_hangup(c);
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
c = NULL;
}
} else { /* Pickup call in call group */
@@ -5961,15 +5959,15 @@
transmit_response_reliable(p, "503 Unavailable", req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
/* Unlock locks so ast_hangup can do its magic */
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
} else {
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
ast_hangup(c);
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
c = NULL;
}
break;
@@ -5998,9 +5996,9 @@
sip_handle_t38_reinvite(bridgepeer, p, 1);
} else { /* Something is wrong with peers udptl struct */
ast_log(LOG_WARNING, "Strange... The other side of the bridge don't have udptl struct\n");
- ast_mutex_lock(&bridgepvt->lock);
+ dialog_lock(bridgepvt, TRUE);
bridgepvt->t38.state = T38_DISABLED;
- ast_mutex_unlock(&bridgepvt->lock);
+ dialog_lock(bridgepvt, FALSE);
if (option_debug > 1)
ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->name);
if (ast_test_flag(req, SIP_PKT_IGNORE))
@@ -6115,18 +6113,6 @@
transmit_response(p, "481 Call Leg Does Not Exist", req);
return 0;
}
-}
-
-/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
-static void stop_media_flows(struct sip_dialog *p)
-{
- /* Immediately stop RTP, VRTP and UDPTL as applicable */
- if (p->rtp)
- ast_rtp_stop(p->rtp);
- if (p->vrtp)
- ast_rtp_stop(p->vrtp);
- if (p->udptl)
- ast_udptl_stop(p->udptl);
}
/*! \brief Handle incoming BYE request */
@@ -6368,7 +6354,7 @@
if (!authpeer || ast_strlen_zero(authpeer->mailbox)) {
transmit_response(p, "404 Not found (no mailbox)", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
+ ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer ? authpeer->name : "<no peername>");
if (authpeer)
ASTOBJ_UNREF(authpeer, sip_destroy_device);
return 0;
@@ -6459,7 +6445,7 @@
continue;
if (p_old->subscribed == NONE)
continue;
- ast_mutex_lock(&p_old->lock);
+ dialog_lock(p_old, TRUE);
if (!strcmp(p_old->peername, p->peername)) {
if (!strcmp(p_old->exten, p->exten) &&
!strcmp(p_old->context, p->context)) {
@@ -6468,7 +6454,7 @@
break;
}
}
- ast_mutex_unlock(&p_old->lock);
+ dialog_lock(p_old, FALSE);
}
dialoglist_unlock();
}
@@ -6822,9 +6808,9 @@
if (!sip->rtptimeout)
return;
while (sip->owner && ast_channel_trylock(sip->owner)) {
- ast_mutex_unlock(&sip->lock);
+ dialog_lock(sip, FALSE);
usleep(1);
- ast_mutex_lock(&sip->lock);
+ dialog_lock(sip, TRUE);
}
if (!sip->owner)
return;
@@ -6893,17 +6879,17 @@
get back to this point every millisecond or less)
*/
for (sip = dialoglist; !fastrestart && sip; sip = sip->next) {
- ast_mutex_lock(&sip->lock);
+ dialog_lock(sip, TRUE);
/* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
check_rtp_timeout(sip, t);
/* If we have sessions that needs to be destroyed, do it now */
if (ast_test_flag(&sip->flags[0], SIP_NEEDDESTROY) && !sip->packets &&
!sip->owner) {
- ast_mutex_unlock(&sip->lock);
+ dialog_lock(sip, FALSE);
__sip_destroy(sip, TRUE, FALSE);
goto restartsearch;
}
- ast_mutex_unlock(&sip->lock);
+ dialog_lock(sip, FALSE);
}
dialoglist_unlock();
@@ -7214,9 +7200,9 @@
printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif
p->prefcodec = oldformat; /* Format for this call */
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
if (!tmpc)
sip_destroy(p);
ast_update_use_count();
@@ -7257,7 +7243,7 @@
return -1;
/* Setup everything on the other side like offered/responded from first side */
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
p->t38.jointcapability = p->t38.peercapability = pvt->t38.jointcapability;
ast_udptl_set_far_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
ast_udptl_set_local_max_datagram(p->udptl, ast_udptl_get_local_max_datagram(pvt->udptl));
@@ -7291,7 +7277,7 @@
}
/* Reset lastrtprx timer */
p->lastrtprx = p->lastrtptx = time(NULL);
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return 0;
} else { /* If we are handling sending 200 OK to the other side of the bridge */
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
@@ -7314,7 +7300,7 @@
}
transmit_response_with_attachment(WITH_T38_SDP, p, "200 OK", &p->initreq, XMIT_CRITICAL);
p->lastrtprx = p->lastrtptx = time(NULL);
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
return 0;
}
}
@@ -7358,7 +7344,7 @@
ast_channel_unlock(chan);
return 0;
}
- ast_mutex_lock(&p->lock);
+ dialog_lock(p, TRUE);
if (!strcasecmp(mode, "info")) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
ast_set_flag(&p->flags[0], SIP_DTMF_INFO);
@@ -7381,7 +7367,7 @@
p->vad = NULL;
}
}
- ast_mutex_unlock(&p->lock);
+ dialog_lock(p, FALSE);
ast_channel_unlock(chan);
return 0;
}
@@ -7699,7 +7685,7 @@
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Session Initiation Protocol (SIP)",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Session Initiation Protocol, chan_sip3 version (SIP)",
.load = load_module,
.unload = unload_module,
.reload = reload,
Modified: team/oej/codename-pineapple/channels/sip3/sip3.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3.h?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3.h Wed Nov 8 16:47:55 2006
@@ -911,7 +911,7 @@
int pokeexpire; /*!< When to expire poke (qualify= checking) */
int lastms; /*!< How long last response took (in ms), or -1 for no response */
int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
- struct timeval ps; /*!< Ping send time */
+ struct timeval ps; /*!< Time when sip_poke_peer() was executed */
struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
struct ast_ha *ha; /*!< Access control list */
Modified: team/oej/codename-pineapple/channels/sip3/sip3_compose.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_compose.c?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_compose.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_compose.c Wed Nov 8 16:47:55 2006
@@ -506,6 +506,7 @@
snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_method2txt(sipmethod));
add_header(req, "Via", p->via);
+ add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
/* SLD: FIXME?: do Route: here too? I think not cos this is the first request.
* OTOH, then we won't have anything in p->route anyway */
/* Build Remote Party-ID and From */
@@ -521,7 +522,6 @@
add_header(req, "Call-ID", p->callid);
add_header(req, "CSeq", tmp);
add_header(req, "User-Agent", global.useragent);
- add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
if (!ast_strlen_zero(p->rpid))
add_header(req, "Remote-Party-ID", p->rpid);
}
@@ -586,6 +586,7 @@
set_destination(p, p->route->hop);
add_route(req, is_strict ? p->route->next : p->route);
}
+ add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
ot = get_header(orig, "To");
of = get_header(orig, "From");
@@ -616,7 +617,6 @@
add_header(req, "CSeq", tmp);
add_header(req, "User-Agent", global.useragent);
- add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
if (!ast_strlen_zero(p->rpid))
add_header(req, "Remote-Party-ID", p->rpid);
Modified: team/oej/codename-pineapple/channels/sip3/sip3_dialog.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_dialog.c?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_dialog.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_dialog.c Wed Nov 8 16:47:55 2006
@@ -134,6 +134,17 @@
ast_log(LOG_DEBUG, "=== SIP dialog list: UNLOCKED\n");
}
+/*! \brief Helper function to lock and unlock, hiding the underlying locking mechanism.
+ \param state TRUE for lock, FALSE for unlock
+*/
+void dialog_lock(struct sip_dialog *dialog, int state)
+{
+ if (state)
+ ast_mutex_lock(&dialog->lock);
+ else
+ ast_mutex_unlock(&dialog->lock);
+}
+
/*! \brief Convert SIP dialog states to string */
const char *dialogstate2str(const enum dialogstate state)
{
@@ -192,7 +203,7 @@
/* Just in case... */
int res = FALSE;
- ast_mutex_lock(&dialog->lock);
+ dialog_lock(dialog, TRUE);
/* Find proper transactoin */
for (cur = dialog->packets; cur; prev = cur, cur = cur->next) {
@@ -217,7 +228,7 @@
break;
}
}
- ast_mutex_unlock(&dialog->lock);
+ dialog_lock(dialog, FALSE);
if (option_debug)
ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", dialog->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
}
@@ -984,7 +995,7 @@
if (found) {
/* Found the call */
- ast_mutex_lock(&cur->lock);
+ dialog_lock(dialog, TRUE);
dialoglist_unlock();
return cur;
}
@@ -1009,7 +1020,7 @@
if ((cur = sip_alloc(req->callid, sin, TRUE, intended_method))) {
/* This method creates dialog */
/* Ok, we've created a dialog, let's go and process it */
- ast_mutex_lock(&cur->lock);
+ dialog_lock(dialog, TRUE);
}
break;
}
Modified: team/oej/codename-pineapple/channels/sip3/sip3_network.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_network.c?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_network.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_network.c Wed Nov 8 16:47:55 2006
@@ -370,8 +370,12 @@
ast_queue_hangup(pkt->owner->owner);
ast_channel_unlock(pkt->owner->owner);
} else {
- /* If no channel owner, destroy now */
- ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);
+ /* If no channel owner, destroy now
+ ...unless it's a SIP options packet, where
+ we want the peerpoke expiry routine handle this.
+ */
+ if (pkt->method != SIP_OPTIONS)
+ ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);
}
}
/* In any case, go ahead and remove the packet */
Modified: team/oej/codename-pineapple/channels/sip3/sip3_refer.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_refer.c?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_refer.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_refer.c Wed Nov 8 16:47:55 2006
@@ -361,8 +361,8 @@
/* Chan2m: The transferer, chan1m: The transferee */
pthread_t th;
- transferee = ast_channel_alloc(0);
- transferer = ast_channel_alloc(0);
+ transferee = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "Parking/%s", chan1->name);
+ transferer = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "SIPPeer/%s", chan2->name);
if ((!transferer) || (!transferee)) {
if (transferee) {
transferee->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
@@ -374,7 +374,6 @@
}
return -1;
}
- ast_string_field_build(transferee, name, "Parking/%s", chan1->name);
/* Make formats okay */
transferee->readformat = chan1->readformat;
@@ -390,7 +389,6 @@
/* We make a clone of the peer channel too, so we can play
back the announcement */
- ast_string_field_build(transferer, name, "SIPPeer/%s", chan2->name);
/* Make formats okay */
transferer->readformat = chan2->readformat;
Modified: team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c Wed Nov 8 16:47:55 2006
@@ -1400,3 +1400,18 @@
ast_mutex_unlock(&p->lock);
return 0;
}
+
+
+/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
+void stop_media_flows(struct sip_dialog *dialog)
+{
+ if (option_debug > 4 && sipdebug)
+ ast_log(LOG_DEBUG, "Stopping media flow for %s\n", dialog->callid);
+ /* Immediately stop RTP, VRTP and UDPTL as applicable */
+ if (dialog->rtp)
+ ast_rtp_stop(dialog->rtp);
+ if (dialog->vrtp)
+ ast_rtp_stop(dialog->vrtp);
+ if (dialog->udptl)
+ ast_udptl_stop(dialog->udptl);
+}
Modified: team/oej/codename-pineapple/channels/sip3/sip3funcs.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3funcs.h?rev=47342&r1=47341&r2=47342&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3funcs.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3funcs.h Wed Nov 8 16:47:55 2006
@@ -183,6 +183,7 @@
GNURK int find_sdp(struct sip_request *req);
GNURK int add_sdp(struct sip_request *resp, struct sip_dialog *p);
GNURK struct ast_frame *sip_read(struct ast_channel *ast);
+GNURK void stop_media_flows(struct sip_dialog *dialog);
/* sip3_config.c */
GNURK void set_device_defaults(struct sip_peer *device);
@@ -222,6 +223,7 @@
GNURK void __sip_destroy(struct sip_dialog *p, int lockowner, int lockdialoglist);
GNURK void __sip_ack(struct sip_dialog *dialog, int seqno, int resp, int sipmethod, int reset);
GNURK int __sip_semi_ack(struct sip_dialog *p, int seqno, int resp, int sipmethod);
+GNURK void dialog_lock(struct sip_dialog *dialog, int state);
/* sip3_services.c - outbound registration for services from other servers/providers */
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