[svn-commits] oej: branch oej/codename-pineapple r46375 - in
/team/oej/codename-pineapple/c...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 28 02:18:20 MST 2006
Author: oej
Date: Sat Oct 28 04:18:18 2006
New Revision: 46375
URL: http://svn.digium.com/view/asterisk?rev=46375&view=rev
Log:
Moving around, moving around...
Added:
team/oej/codename-pineapple/channels/sip3/sip3_services.c (with props)
Modified:
team/oej/codename-pineapple/channels/Makefile
team/oej/codename-pineapple/channels/chan_sip3.c
team/oej/codename-pineapple/channels/sip3/Makefile
team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c
team/oej/codename-pineapple/channels/sip3/sip3funcs.h
Modified: team/oej/codename-pineapple/channels/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/Makefile?rev=46375&r1=46374&r2=46375&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/Makefile (original)
+++ team/oej/codename-pineapple/channels/Makefile Sat Oct 28 04:18:18 2006
@@ -15,7 +15,7 @@
CC_MODS:=$(filter-out $(MENUSELECT_CHANNELS),$(patsubst %.cc,%,$(wildcard chan_*.cc)))
SIP3_MODULES=chan_sip3.o sip3/sip3_network.o sip3/sip3_subscribe.o sip3/sip3_refer.o sip3/sip3_domain.o \
sip3/sip3_callerid.o sip3/sip3_auth.o sip3/sip3_sdprtp.o sip3/sip3_config.o \
- sip3/sip3_cliami.o sip3/sip3_dialog.o
+ sip3/sip3_cliami.o sip3/sip3_dialog.o sip3/sip3_services.o
ifeq ($(OSARCH),OpenBSD)
PTLIB=-lpt_OpenBSD_x86_r
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=46375&r1=46374&r2=46375&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Sat Oct 28 04:18:18 2006
@@ -232,6 +232,7 @@
- \b sip3_parse.c Parsing stuff
- \b sip3_refer.c SIP transfer support
- \b sip3_network.c Networks interface (UDP today)
+ - \b sip3_services.c Outbound registrations (services)
*/
/* -END- documentation pages */
@@ -583,14 +584,12 @@
static void handle_response(struct sip_dialog *p, int resp, char *rest, struct sip_request *req, int seqno);
/*----- RTP interface functions */
-static int sip_get_codec(struct ast_channel *chan);
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_dialog *p, int *faxdetect);
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 */
static int transmit_response_with_t38_sdp(struct sip_dialog *p, char *msg, struct sip_request *req, int retrans);
-static int transmit_reinvite_with_t38_sdp(struct sip_dialog *p);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
@@ -601,38 +600,22 @@
.description = "Session Initiation Protocol (SIP)",
.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
- .requester = sip_request_call,
- .devicestate = sip_devicestate,
- .call = sip_call,
- .hangup = sip_hangup,
- .answer = sip_answer,
- .read = sip_read,
- .write = sip_write,
- .write_video = sip_write,
- .indicate = sip_indicate,
- .transfer = sip_transfer,
+ .requester = sip_request_call, /*!< Where we set up a call, but don't actually activate it */
+ .devicestate = sip_devicestate, /*!< Checking the status of a known SIP device */
+ .call = sip_call, /*!< Try calling Bob, says Alice */
+ .hangup = sip_hangup, /*!< Alice does not want to talk to Bob any more */
+ .answer = sip_answer, /*!< Bob answers the call */
+ .read = sip_read, /*!< Deliver media to the PBX */
+ .write = sip_write, /*!< Get media from the PBX side */
+ .write_video = sip_write, /*!< Get video media from the PBX side */
+ .indicate = sip_indicate, /*!< Get indications from the PBX side */
+ .transfer = sip_transfer, /*!< Transfer a call, severely broken */
.fixup = sip_fixup,
- .send_digit_begin = sip_senddigit_begin,
- .send_digit_end = sip_senddigit_end,
+ .send_digit_begin = sip_senddigit_begin, /*!< DTMF support */
+ .send_digit_end = sip_senddigit_end, /*!< DTMF support */
.bridge = ast_rtp_bridge,
.early_bridge = ast_rtp_early_bridge,
- .send_text = sip_sendtext,
-};
-
-/*! \brief Interface structure with callbacks used to connect to RTP module */
-static struct ast_rtp_protocol sip_rtp = {
- type: "SIP",
- get_rtp_info: sip_get_rtp_peer,
- get_vrtp_info: sip_get_vrtp_peer,
- set_rtp_peer: sip_set_rtp_peer,
- get_codec: sip_get_codec,
-};
-
-/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
-static struct ast_udptl_protocol sip_udptl = {
- type: "SIP",
- get_udptl_info: sip_get_udptl_peer,
- set_udptl_peer: sip_set_udptl_peer,
+ .send_text = sip_sendtext, /*!< Get text from the PBX to send out */
};
/*! \brief Initialize the initital request packet in the pvt structure.
@@ -2526,103 +2509,6 @@
return __get_header(req, name, &start);
}
-/*! \brief Read RTP from network */
-static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_dialog *p, int *faxdetect)
-{
- /* Retrieve audio/etc from channel. Assumes p->lock is already held. */
- struct ast_frame *f;
-
- if (!p->rtp) {
- /* We have no RTP allocated for this channel */
- return &ast_null_frame;
- }
-
- switch(ast->fdno) {
- case 0:
- f = ast_rtp_read(p->rtp); /* RTP Audio */
- break;
- case 1:
- f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */
- break;
- case 2:
- f = ast_rtp_read(p->vrtp); /* RTP Video */
- break;
- case 3:
- f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */
- break;
- case 5:
- f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
- break;
- default:
- f = &ast_null_frame;
- }
- /* Don't forward RFC2833 if we're not supposed to */
- if (f && (f->frametype == AST_FRAME_DTMF) &&
- (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833))
- return &ast_null_frame;
-
- if (p->owner) {
- /* We already hold the channel lock */
- if (f->frametype == AST_FRAME_VOICE) {
- if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
- if (option_debug)
- ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
- p->owner->nativeformats = (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass;
- ast_set_read_format(p->owner, p->owner->readformat);
- ast_set_write_format(p->owner, p->owner->writeformat);
- }
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
- f = ast_dsp_process(p->owner, p->vad, f);
- if (f && f->frametype == AST_FRAME_DTMF) {
- if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
- if (option_debug)
- ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
- *faxdetect = 1;
- } else if (option_debug) {
- ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass);
- }
- }
- }
- }
- }
- return f;
-}
-
-/*! \brief Read SIP RTP from channel */
-static struct ast_frame *sip_read(struct ast_channel *ast)
-{
- struct ast_frame *fr;
- struct sip_dialog *p = ast->tech_pvt;
- int faxdetected = FALSE;
-
- ast_mutex_lock(&p->lock);
- fr = sip_rtp_read(ast, p, &faxdetected);
- p->lastrtprx = time(NULL);
-
- /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
- /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
- if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
- if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- if (!p->pendinginvite) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
- p->t38.state = T38_LOCAL_REINVITE;
- transmit_reinvite_with_t38_sdp(p);
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name);
- }
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
-
- ast_mutex_unlock(&p->lock);
- return fr;
-}
-
-
/*! \brief Generate 32 byte random string for callid's etc */
static char *generate_random_string(char *buf, size_t size)
{
@@ -3957,7 +3843,7 @@
We reinvite so that the T38 processing can take place.
SIP Signalling stays with * in the path.
*/
-static int transmit_reinvite_with_t38_sdp(struct sip_dialog *p)
+GNURK int transmit_reinvite_with_t38_sdp(struct sip_dialog *p)
{
struct sip_request req;
@@ -10650,53 +10536,6 @@
return peer;
}
-static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
-{
- struct sip_dialog *p;
- struct ast_udptl *udptl = NULL;
-
- p = chan->tech_pvt;
- if (!p)
- return NULL;
-
- ast_mutex_lock(&p->lock);
- if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
- udptl = p->udptl;
- ast_mutex_unlock(&p->lock);
- return udptl;
-}
-
-static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
-{
- struct sip_dialog *p;
-
- p = chan->tech_pvt;
- if (!p)
- return -1;
- ast_mutex_lock(&p->lock);
- if (udptl)
- ast_udptl_get_peer(udptl, &p->udptlredirip);
- else
- memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
- if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- if (!p->pendinginvite) {
- if (option_debug > 2) {
- ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
- }
- transmit_reinvite_with_t38_sdp(p);
- } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- if (option_debug > 2) {
- ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
- }
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
- }
- /* Reset lastrtprx timer */
- p->lastrtprx = p->lastrtptx = time(NULL);
- ast_mutex_unlock(&p->lock);
- return 0;
-}
-
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_dialog *pvt, int reinvite)
{
struct sip_dialog *p;
@@ -10930,13 +10769,6 @@
return -1;
}
-/*! \brief Return SIP UA's codec (part of the RTP interface) */
-static int sip_get_codec(struct ast_channel *chan)
-{
- struct sip_dialog *p = chan->tech_pvt;
- return p->peercapability;
-}
-
/*! \brief Send a poke to all known peers
Space them out 100 ms apart
XXX We might have a cool algorithm for this or use random - any suggestions?
@@ -11085,11 +10917,8 @@
return AST_MODULE_LOAD_FAILURE;
}
- /* Tell the RTP subdriver that we're here */
- ast_rtp_proto_register(&sip_rtp);
-
- /* Tell the UDPTL subdriver that we're here */
- ast_udptl_proto_register(&sip_udptl);
+ /* Tell the RTP and UDPTL subdriver that we're here */
+ register_rtp_and_udptl(); /* See sip3_sdprtp.c */
/* Register dialplan applications */
ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
@@ -11128,20 +10957,20 @@
ast_unregister_application(app_sipaddheader);
sip_cli_and_manager_commands_unregister();
- ast_rtp_proto_unregister(&sip_rtp);
- ast_udptl_proto_unregister(&sip_udptl);
+ unregister_rtp_and_udptl(); /* sip3_sdprtp.c */
ast_manager_unregister("SIPpeers");
ast_manager_unregister("SIPshowpeer");
+ /* Hangup all dialogs if they have an owner */
dialoglist_lock();
- /* Hangup all interfaces if they have an owner */
for (p = dialoglist; p ; p = p->next) {
if (p->owner)
ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
}
dialoglist_unlock();
+ /* Kill the monitor thread */
ast_mutex_lock(&monlock);
if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP)) {
pthread_cancel(monitor_thread);
@@ -11151,8 +10980,8 @@
monitor_thread = AST_PTHREADT_STOP;
ast_mutex_unlock(&monlock);
+ /* Destroy all the dialogs and free their memory */
dialoglist_lock();
- /* Destroy all the interfaces and free their memory */
p = dialoglist;
while (p) {
pl = p;
@@ -11171,14 +11000,18 @@
/* Free memory for local network address mask */
ast_free_ha(sipnet.localaddr);
+ /* Destroy all device configurations in the device list */
ASTOBJ_CONTAINER_DESTROYALL(&devicelist, sip_destroy_device);
ASTOBJ_CONTAINER_DESTROY(&devicelist);
+
+ /* Destroy all entries in the registry list */
ASTOBJ_CONTAINER_DESTROYALL(®l, sip_registry_destroy);
ASTOBJ_CONTAINER_DESTROY(®l);
clear_realm_authentication(authl);
- clear_sip_domains();
- close(sipnet.sipsock);
+ clear_sip_domains(); /* Clear the list of hosted domains */
+
+ close(sipnet.sipsock); /* Close the network socket */
sched_context_destroy(sched);
return 0;
Modified: team/oej/codename-pineapple/channels/sip3/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/Makefile?rev=46375&r1=46374&r2=46375&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/Makefile (original)
+++ team/oej/codename-pineapple/channels/sip3/Makefile Sat Oct 28 04:18:18 2006
@@ -2,7 +2,7 @@
# Makefile for chan_sip3 support
#
all: sip3_refer.o sip3_subscribe.o sip3_network.o sip3_domain.o sip3_callerid.o sip3_auth.o \
- sip3_sdprtp.o sip3_config.o sip3_cliami.c sip3_dialog.c
+ sip3_sdprtp.o sip3_config.o sip3_cliami.o sip3_dialog.o sip3_services.o
%.o: %.c sip3funcs.h sip3.h
$(CC) $(CFLAGS) -c -o $@ $<
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=46375&r1=46374&r2=46375&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c Sat Oct 28 04:18:18 2006
@@ -84,6 +84,48 @@
#include "asterisk/compiler.h"
#include "sip3.h"
#include "sip3funcs.h"
+
+/* Forward declarations */
+static int sip_get_codec(struct ast_channel *chan);
+static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
+static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
+
+/*! \brief Interface structure with callbacks used to connect to RTP module */
+static struct ast_rtp_protocol sip_rtp = {
+ type: "SIP",
+ get_rtp_info: sip_get_rtp_peer,
+ get_vrtp_info: sip_get_vrtp_peer,
+ set_rtp_peer: sip_set_rtp_peer,
+ get_codec: sip_get_codec,
+};
+
+/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
+static struct ast_udptl_protocol sip_udptl = {
+ type: "SIP",
+ get_udptl_info: sip_get_udptl_peer,
+ set_udptl_peer: sip_set_udptl_peer,
+};
+
+/*! \brief Register RTP and UDPTL to the subsystems */
+void register_rtp_and_udptl(void)
+{
+ /* Tell the RTP subdriver that we're here */
+ ast_rtp_proto_register(&sip_rtp);
+
+ /* Tell the UDPTL subdriver that we're here */
+ ast_udptl_proto_register(&sip_udptl);
+}
+
+/*! \brief UNRegister RTP and UDPTL to the subsystems */
+void unregister_rtp_and_udptl(void)
+{
+ /* Tell the RTP subdriver that we're gone */
+ ast_rtp_proto_unregister(&sip_rtp);
+
+ /* Tell the UDPTL subdriver that we're gone */
+ ast_udptl_proto_unregister(&sip_udptl);
+}
+
/*! \brief Reads one line of SIP message body */
static char *get_body_by_line(const char *line, const char *name, int nameLen)
@@ -1206,3 +1248,154 @@
return 0;
}
+/*! \brief Return SIP UA's codec (part of the RTP interface) */
+static int sip_get_codec(struct ast_channel *chan)
+{
+ struct sip_dialog *p = chan->tech_pvt;
+ return p->peercapability;
+}
+
+/*! \brief Read RTP from network */
+static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_dialog *p, int *faxdetect)
+{
+ /* Retrieve audio/etc from channel. Assumes p->lock is already held. */
+ struct ast_frame *f;
+
+ if (!p->rtp) {
+ /* We have no RTP allocated for this channel */
+ return &ast_null_frame;
+ }
+
+ switch(ast->fdno) {
+ case 0:
+ f = ast_rtp_read(p->rtp); /* RTP Audio */
+ break;
+ case 1:
+ f = ast_rtcp_read(p->rtp); /* RTCP Control Channel */
+ break;
+ case 2:
+ f = ast_rtp_read(p->vrtp); /* RTP Video */
+ break;
+ case 3:
+ f = ast_rtcp_read(p->vrtp); /* RTCP Control Channel for video */
+ break;
+ case 5:
+ f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
+ break;
+ default:
+ f = &ast_null_frame;
+ }
+ /* Don't forward RFC2833 if we're not supposed to */
+ if (f && (f->frametype == AST_FRAME_DTMF) &&
+ (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833))
+ return &ast_null_frame;
+
+ if (p->owner) {
+ /* We already hold the channel lock */
+ if (f->frametype == AST_FRAME_VOICE) {
+ if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
+ p->owner->nativeformats = (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass;
+ ast_set_read_format(p->owner, p->owner->readformat);
+ ast_set_write_format(p->owner, p->owner->writeformat);
+ }
+ if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
+ f = ast_dsp_process(p->owner, p->vad, f);
+ if (f && f->frametype == AST_FRAME_DTMF) {
+ if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
+ *faxdetect = 1;
+ } else if (option_debug) {
+ ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass);
+ }
+ }
+ }
+ }
+ }
+ return f;
+}
+
+/*! \brief Read SIP RTP from channel */
+static struct ast_frame *sip_read(struct ast_channel *ast)
+{
+ struct ast_frame *fr;
+ struct sip_dialog *p = ast->tech_pvt;
+ int faxdetected = FALSE;
+
+ ast_mutex_lock(&p->lock);
+ fr = sip_rtp_read(ast, p, &faxdetected);
+ p->lastrtprx = time(NULL);
+
+ /* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
+ /* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
+ if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
+ if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+ if (!p->pendinginvite) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
+ p->t38.state = T38_LOCAL_REINVITE;
+ transmit_reinvite_with_t38_sdp(p);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name);
+ }
+ } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
+ }
+
+ ast_mutex_unlock(&p->lock);
+ return fr;
+}
+
+/*! \brief Get UDPTL peer address (part of UDPTL interface) */
+static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
+{
+ struct sip_dialog *p;
+ struct ast_udptl *udptl = NULL;
+
+ p = chan->tech_pvt;
+ if (!p)
+ return NULL;
+
+ ast_mutex_lock(&p->lock);
+ if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
+ udptl = p->udptl;
+ ast_mutex_unlock(&p->lock);
+ return udptl;
+}
+
+/*! \brief Determine UDPTL peer address for re-invite (part of UDPTL interface) */
+static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
+{
+ struct sip_dialog *p;
+
+ p = chan->tech_pvt;
+ if (!p)
+ return -1;
+ ast_mutex_lock(&p->lock);
+ if (udptl)
+ ast_udptl_get_peer(udptl, &p->udptlredirip);
+ else
+ memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
+ if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+ if (!p->pendinginvite) {
+ if (option_debug > 2) {
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
+ }
+ transmit_reinvite_with_t38_sdp(p);
+ } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
+ if (option_debug > 2) {
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
+ }
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
+ }
+ /* Reset lastrtprx timer */
+ p->lastrtprx = p->lastrtptx = time(NULL);
+ ast_mutex_unlock(&p->lock);
+ return 0;
+}
Added: team/oej/codename-pineapple/channels/sip3/sip3_services.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_services.c?rev=46375&view=auto
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_services.c (added)
+++ team/oej/codename-pineapple/channels/sip3/sip3_services.c Sat Oct 28 04:18:18 2006
@@ -1,0 +1,94 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2006, Digium, Inc.
+ *
+ * Mark Spencer <markster at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Various SIP functions for services, registrations for SIP service with other providers/servers
+ * Version 3 of chan_sip
+ *
+ * \author Mark Spencer <markster at digium.com>
+ * \author Olle E. Johansson <oej at edvina.net> (all the chan_sip3 changes)
+ *
+ * See Also:
+ * \arg \ref AstCREDITS
+ * \arg \ref Chan_sip3_00index
+ *
+ *
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <stdio.h>
+#include <ctype.h>
+#include <string.h>
+#include <unistd.h>
+#include <sys/socket.h>
+#include <sys/ioctl.h>
+#include <net/if.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <netdb.h>
+#include <signal.h>
+#include <sys/signal.h>
+#include <netinet/in.h>
+#include <netinet/in_systm.h>
+#include <arpa/inet.h>
+#include <netinet/ip.h>
+#include <regex.h>
+
+#include "asterisk/lock.h"
+#include "asterisk/channel.h"
+#include "asterisk/config.h"
+#include "asterisk/logger.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/lock.h"
+#include "asterisk/sched.h"
+#include "asterisk/io.h"
+#include "asterisk/rtp.h"
+#include "asterisk/udptl.h"
+#include "asterisk/acl.h"
+#include "asterisk/manager.h"
+#include "asterisk/callerid.h"
+#include "asterisk/cli.h"
+#include "asterisk/app.h"
+#include "asterisk/musiconhold.h"
+#include "asterisk/dsp.h"
+#include "asterisk/features.h"
+#include "asterisk/acl.h"
+#include "asterisk/srv.h"
+#include "asterisk/astdb.h"
+#include "asterisk/causes.h"
+#include "asterisk/utils.h"
+#include "asterisk/file.h"
+#include "asterisk/astobj.h"
+#include "asterisk/dnsmgr.h"
+#include "asterisk/devicestate.h"
+#include "asterisk/linkedlists.h"
+#include "asterisk/stringfields.h"
+#include "asterisk/monitor.h"
+#include "asterisk/localtime.h"
+#include "asterisk/abstract_jb.h"
+#include "asterisk/compiler.h"
+#include "sip3.h"
+#include "sip3funcs.h"
+
Propchange: team/oej/codename-pineapple/channels/sip3/sip3_services.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/oej/codename-pineapple/channels/sip3/sip3_services.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/oej/codename-pineapple/channels/sip3/sip3_services.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
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=46375&r1=46374&r2=46375&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3funcs.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3funcs.h Sat Oct 28 04:18:18 2006
@@ -78,6 +78,7 @@
GNURK int transmit_response_with_auth(struct sip_dialog *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
GNURK int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
GNURK int transmit_invite(struct sip_dialog *p, int sipmethod, int sdp, int init);
+GNURK int transmit_reinvite_with_t38_sdp(struct sip_dialog *p);
GNURK void sip_scheddestroy(struct sip_dialog *p, int ms);
/*! sip3_refer.c */
@@ -131,6 +132,8 @@
GNURK int build_reply_digest(struct sip_dialog *p, int method, char* digest, int digest_len);
/* sip3_sdprtp.c */
+GNURK void register_rtp_and_udptl(void);
+GNURK void unregister_rtp_and_udptl(void);
GNURK char *get_body(struct sip_request *req, char *name);
GNURK int process_sdp(struct sip_dialog *p, struct sip_request *req);
GNURK int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
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