[svn-commits] oej: branch oej/codename-pineapple r45763 - in
/team/oej/codename-pineapple/c...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Fri Oct 20 08:31:27 MST 2006
Author: oej
Date: Fri Oct 20 10:31:27 2006
New Revision: 45763
URL: http://svn.digium.com/view/asterisk?rev=45763&view=rev
Log:
Staying on track with trunk...
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
team/oej/codename-pineapple/channels/sip3/sip3.h
team/oej/codename-pineapple/channels/sip3/sip3_auth.c
team/oej/codename-pineapple/channels/sip3/sip3_callerid.c
team/oej/codename-pineapple/channels/sip3/sip3_network.c
team/oej/codename-pineapple/channels/sip3/sip3_parse.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=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Fri Oct 20 10:31:27 2006
@@ -10,6 +10,7 @@
* Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster at digium.com>
+ * Chan_sip3 changes by Olle E. Johansson <oej at edvina.net>
*
*
*
@@ -263,6 +264,10 @@
/*! \brief The register list: Other SIP proxys we register with and place calls to */
struct sip_register_list regl;
+/* Forward declaration */
+static int temp_pvt_init(void *data);
+static void temp_pvt_cleanup(void *data);
+
/*! \brief A per-thread temporary pvt structure */
AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
@@ -350,15 +355,10 @@
/*--- Authentication stuff */
static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
-static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */
static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
int sipmethod, char *uri, enum xmittype reliable,
struct sockaddr_in *sin, struct sip_peer **authpeer);
static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
-
-/*--- SIP realm authentication */
-static int clear_realm_authentication(struct sip_auth *authlist);
-static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
/*--- Misc functions */
static int sip_do_reload(enum channelreloadreason reason);
@@ -480,8 +480,6 @@
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
static int set_address_from_contact(struct sip_pvt *pvt);
static void check_via(struct sip_pvt *p, struct sip_request *req);
-static char *get_calleridname(const char *input, char *output, size_t outputsize);
-static int get_rpid_num(const char *input, char *output, int maxlen);
static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
static int get_msg_text(char *buf, int len, struct sip_request *req);
@@ -504,7 +502,6 @@
static void build_callid_pvt(struct sip_pvt *pvt);
static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
static void make_our_tag(char *tagbuf, size_t len);
-static int add_line(struct sip_request *req, const char *line);
static int add_text(struct sip_request *req, const char *text);
static int add_digit(struct sip_request *req, char digit);
static int add_vidupdate(struct sip_request *req);
@@ -542,6 +539,7 @@
/*----- 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_pvt *p, int *faxdetect);
+static void stop_media_flows(struct sip_pvt *p);
/*------ T38 Support --------- */
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
@@ -717,7 +715,7 @@
{
va_list ap;
- if (!global.recordhistory || !p)
+ if (!p)
return;
va_start(ap, fmt);
append_history_va(p, fmt, ap);
@@ -768,7 +766,7 @@
}
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_method2txt(p->method));
- if (global.recordhistory)
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "SchedDestroy", "%d ms", ms);
if (p->autokillid > -1)
@@ -1440,7 +1438,7 @@
ast_sched_del(sched, reg->expire);
if (reg->timeout > -1)
ast_sched_del(sched, reg->timeout);
- ast_string_field_free_all(reg);
+ ast_string_field_free_pools(reg);
sipcounters.registry_objects--;
free(reg);
@@ -1533,7 +1531,7 @@
}
ast_mutex_destroy(&p->lock);
- ast_string_field_free_all(p);
+ ast_string_field_free_pools(p);
free(p);
}
@@ -1932,7 +1930,7 @@
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
/* Get RTCP quality before end of call */
- if (global.recordhistory) {
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
if (p->rtp)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
@@ -2279,7 +2277,7 @@
and from handle_request_invite for inbound channels
*/
-static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title)
+static struct ast_channel *sip_new(struct sip_pvt *dialog, int state, const char *title)
{
struct ast_channel *tmp;
struct ast_variable *v = NULL;
@@ -2287,10 +2285,10 @@
int what;
int needvideo = 0;
- ast_mutex_unlock(&i->lock);
+ ast_mutex_unlock(&dialog->lock);
/* Don't hold a sip pvt lock while we allocate a channel */
tmp = ast_channel_alloc(1);
- ast_mutex_lock(&i->lock);
+ ast_mutex_lock(&dialog->lock);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
return NULL;
@@ -2299,23 +2297,23 @@
/* Select our native format based on codec preference until we receive
something from another device to the contrary. */
- if (i->jointcapability) /* The joint capabilities of us and peer */
- what = i->jointcapability;
- else if (i->capability) /* Our configured capability for this peer */
- what = i->capability;
+ if (dialog->jointcapability) /* The joint capabilities of us and peer */
+ what = dialog->jointcapability;
+ else if (dialog->capability) /* Our configured capability for this peer */
+ what = dialog->capability;
else
what = global.capability; /* Global codec support */
/* Set the native formats for audio and merge in video */
- tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
+ tmp->nativeformats = ast_codec_choose(&dialog->prefs, what, 1) | (dialog->jointcapability & AST_FORMAT_VIDEO_MASK);
if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
- ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
- ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
- ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1)));
- if (i->prefcodec)
- ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec));
+ ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, dialog->jointcapability));
+ ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, dialog->capability));
+ ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&dialog->prefs, what, 1)));
+ if (dialog->prefcodec)
+ ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, dialog->prefcodec));
}
/* XXX Why are we choosing a codec from the native formats?? */
@@ -2325,11 +2323,11 @@
preferred format for this call. Otherwise, we check the jointcapability
We also check for vrtp. If it's not there, we are not allowed do any video anyway.
*/
- if (i->vrtp) {
- if (i->prefcodec)
- needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */
+ if (dialog->vrtp) {
+ if (dialog->prefcodec)
+ needvideo = dialog->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */
else
- needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */
+ needvideo = dialog->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */
}
if (option_debug > 2) {
@@ -2344,29 +2342,29 @@
const char *my_name; /* pick a good name */
if (title)
my_name = title;
- else if ( (my_name = strchr(i->fromdomain,':')) )
+ else if ( (my_name = strchr(dialog->fromdomain,':')) )
my_name++; /* skip ':' */
else
- my_name = i->fromdomain;
- ast_string_field_build(tmp, name, "SIP/%s-%08x", my_name, (int)(long) i);
- }
-
- if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
- i->vad = ast_dsp_new();
- ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT);
+ 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) {
+ dialog->vad = ast_dsp_new();
+ ast_dsp_set_features(dialog->vad, DSP_FEATURE_DTMF_DETECT);
if (global.relaxdtmf)
- ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
- }
- if (i->rtp) {
- tmp->fds[0] = ast_rtp_fd(i->rtp);
- tmp->fds[1] = ast_rtcp_fd(i->rtp);
- }
- if (needvideo && i->vrtp) {
- tmp->fds[2] = ast_rtp_fd(i->vrtp);
- tmp->fds[3] = ast_rtcp_fd(i->vrtp);
- }
- if (i->udptl) {
- tmp->fds[5] = ast_udptl_fd(i->udptl);
+ ast_dsp_digitmode(dialog->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
+ }
+ if (dialog->rtp) {
+ tmp->fds[0] = ast_rtp_fd(dialog->rtp);
+ tmp->fds[1] = ast_rtcp_fd(dialog->rtp);
+ }
+ if (needvideo && dialog->vrtp) {
+ tmp->fds[2] = ast_rtp_fd(dialog->vrtp);
+ tmp->fds[3] = ast_rtcp_fd(dialog->vrtp);
+ }
+ if (dialog->udptl) {
+ tmp->fds[5] = ast_udptl_fd(dialog->udptl);
}
if (state == AST_STATE_RING)
tmp->rings = 1;
@@ -2375,43 +2373,43 @@
tmp->rawwriteformat = fmt;
tmp->readformat = fmt;
tmp->rawreadformat = fmt;
- tmp->tech_pvt = i;
-
- tmp->callgroup = i->callgroup;
- tmp->pickupgroup = i->pickupgroup;
- tmp->cid.cid_pres = i->callingpres;
- if (!ast_strlen_zero(i->accountcode))
- ast_string_field_set(tmp, accountcode, i->accountcode);
- if (i->amaflags)
- tmp->amaflags = i->amaflags;
- if (!ast_strlen_zero(i->language))
- ast_string_field_set(tmp, language, i->language);
- i->owner = tmp;
+ tmp->tech_pvt = dialog;
+
+ tmp->callgroup = dialog->callgroup;
+ tmp->pickupgroup = dialog->pickupgroup;
+ tmp->cid.cid_pres = dialog->callingpres;
+ if (!ast_strlen_zero(dialog->accountcode))
+ ast_string_field_set(tmp, accountcode, dialog->accountcode);
+ if (dialog->amaflags)
+ tmp->amaflags = dialog->amaflags;
+ if (!ast_strlen_zero(dialog->language))
+ ast_string_field_set(tmp, language, dialog->language);
+ dialog->owner = tmp;
ast_atomic_fetchadd_int(&usecnt, 1);
ast_update_use_count();
- ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
- ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
+ ast_copy_string(tmp->context, dialog->context, sizeof(tmp->context));
+ ast_copy_string(tmp->exten, dialog->exten, sizeof(tmp->exten));
/* Don't use ast_set_callerid() here because it will
* generate a NewCallerID event before the NewChannel event */
- tmp->cid.cid_num = ast_strdup(i->cid_num);
- tmp->cid.cid_ani = ast_strdup(i->cid_num);
- tmp->cid.cid_name = ast_strdup(i->cid_name);
- if (!ast_strlen_zero(i->rdnis))
- tmp->cid.cid_rdnis = ast_strdup(i->rdnis);
+ tmp->cid.cid_num = ast_strdup(dialog->cid_num);
+ tmp->cid.cid_ani = ast_strdup(dialog->cid_num);
+ tmp->cid.cid_name = ast_strdup(dialog->cid_name);
+ if (!ast_strlen_zero(dialog->rdnis))
+ tmp->cid.cid_rdnis = ast_strdup(dialog->rdnis);
- if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
- tmp->cid.cid_dnid = ast_strdup(i->exten);
+ if (!ast_strlen_zero(dialog->exten) && strcmp(dialog->exten, "s"))
+ tmp->cid.cid_dnid = ast_strdup(dialog->exten);
tmp->priority = 1;
- if (!ast_strlen_zero(i->uri))
- pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
- if (!ast_strlen_zero(i->domain))
- pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
- if (!ast_strlen_zero(i->callid))
- pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
+ if (!ast_strlen_zero(dialog->uri))
+ pbx_builtin_setvar_helper(tmp, "SIPURI", dialog->uri);
+ if (!ast_strlen_zero(dialog->domain))
+ 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 (i->rtp)
+ if (dialog->rtp)
ast_jb_configure(tmp, &global.jbconf);
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
@@ -2420,11 +2418,11 @@
tmp = NULL;
}
/* Set channel variables for this call from configuration */
- for (v = i->chanvars ; v ; v = v->next)
+ for (v = dialog->chanvars ; v ; v = v->next)
pbx_builtin_setvar_helper(tmp,v->name,v->value);
- if (global.recordhistory)
- append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
+ if (!ast_test_flag(&dialog->flags[0], SIP_NO_HISTORY))
+ append_history(dialog, "NewChan", "Channel %s - from %s", tmp->name, dialog->callid);
return tmp;
}
@@ -2678,6 +2676,8 @@
ast_copy_flags(&p->flags[0], &global.flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &global.flags[1], SIP_PAGE2_FLAGS_TO_COPY);
+ ast_set2_flag(&p->flags[0], !global.recordhistory, SIP_NO_HISTORY);
+
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
p->ocseq = INITIAL_CSEQ;
@@ -2827,19 +2827,19 @@
}
ast_mutex_unlock(&iflock);
if (sip_methods[intended_method].creates_dialog == CAN_CREATE_DIALOG) {
- /* This method creates dialog */
- if ((p = sip_alloc(callid, sin, 1, intended_method)))
+ if (intended_method == SIP_REFER) {
+
+ /* We do not support out-of-dialog REFERs yet */
+ transmit_response_using_temp(callid, sin, 1, intended_method, req, "603 Declined (no dialog)");
+ } else if (intended_method == SIP_NOTIFY) {
+ /* We do not support out-of-dialog NOTIFY either,
+ like voicemail notification, so cancel that early */
+ transmit_response_using_temp(callid, sin, 1, intended_method, req, "489 Bad event");
+ } else if ((p = sip_alloc(callid, sin, 1, intended_method))) {
+ /* This method creates dialog */
/* Ok, we've created a dialog, let's go and process it */
ast_mutex_lock(&p->lock);
- return(p);
- } else if (intented_method = SIP_REFER) {
-
- /* We do not support out-of-dialog REFERs yet */
- transmit_response_using_temp(callid, sin, 1, intended_method, req, "603 Declined (no dialog)");
- } else if (intented_method = SIP_NOTIFY) {
- /* We do not support out-of-dialog NOTIFY either,
- like voicemail notification, so cancel that early */
- transmit_response_using_temp(callid, sin, 1, intended_method, req, "489 Bad event");
+ }
} else {
if (intended_method != SIP_RESPONSE)
transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist");
@@ -3062,7 +3062,7 @@
}
/*! \brief Add content (not header) to SIP message */
-static int add_line(struct sip_request *req, const char *line)
+GNURK int add_line(struct sip_request *req, const char *line)
{
if (req->lines == SIP_MAX_LINES) {
ast_log(LOG_WARNING, "Out of SIP line space\n");
@@ -3503,27 +3503,29 @@
return -1;
}
- memset(p, 0, sizeof(*p));
+ /* if the structure was just allocated, initialize it */
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
+ ast_set_flag(&p->flags[0], SIP_NO_HISTORY);
+ if (ast_string_field_init(p, 512))
+ return -1;
+ }
/* Initialize the bare minimum */
- if (ast_string_field_init(p, 512))
- return -1;
-
p->method = intended_method;
if (sin) {
p->sa = *sin;
if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
- p->ourip = __ourip;
+ p->ourip = sipnet.__ourip;
} else
- p->ourip = __ourip;
+ p->ourip = sipnet.__ourip;
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
p->ocseq = INITIAL_CSEQ;
if (useglobal_nat && sin) {
- ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
+ ast_copy_flags(&p->flags[0], &global.flags[0], SIP_NAT);
p->recv = *sin;
do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
}
@@ -3535,7 +3537,7 @@
/* Use this temporary pvt structure to send the message */
__transmit_response(p, msg, req, XMIT_UNRELIABLE);
- /* Now do a simple destruction */
+ /* Free the string fields, but not the pool space */
ast_string_field_free_all(p);
return 0;
@@ -3912,7 +3914,7 @@
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug)
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
- if (global.recordhistory)
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent");
add_sdp(&req, p);
/* Use this as the basis */
@@ -4538,7 +4540,7 @@
if (!r)
return 0;
- if (r->call && global.recordhistory)
+ if (r->call && !ast_test_flag(&r->call->flags[0], SIP_NO_HISTORY))
append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
/* Since registry's are only added/removed by the the monitor thread, this
may be overkill to reference/dereference at all here */
@@ -4638,7 +4640,7 @@
ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
return 0;
}
- if (global.recordhistory)
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
/* Find address to hostname */
if (create_addr(p, r->hostname)) {
@@ -9047,18 +9049,7 @@
if ((option_verbose > 2) && (resp != 487))
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
- if (p->rtp) {
- /* Immediately stop RTP */
- ast_rtp_stop(p->rtp);
- }
- if (p->vrtp) {
- /* Immediately stop VRTP */
- ast_rtp_stop(p->vrtp);
- }
- if (p->udptl) {
- /* Immediately stop UDPTL */
- ast_udptl_stop(p->udptl);
- }
+ stop_media_flows(p); /* Stop RTP, VRTP and UDPTL */
/* XXX Locking issues?? XXX */
switch(resp) {
case 300: /* Multiple Choices */
@@ -10022,7 +10013,7 @@
if (option_debug)
ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
}
- if (global.recordhistory) /* This is a response, note what it was for */
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a response, note what it was for */
append_history(p, "ReInv", "Re-invite received");
}
} else if (debug)
@@ -10790,18 +10781,7 @@
ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n");
return 0;
}
- if (p->rtp) {
- /* Immediately stop RTP */
- ast_rtp_stop(p->rtp);
- }
- if (p->vrtp) {
- /* Immediately stop VRTP */
- ast_rtp_stop(p->vrtp);
- }
- if (p->udptl) {
- /* Immediately stop UDPTL */
- ast_udptl_stop(p->udptl);
- }
+ stop_media_flows(p);
if (p->owner)
ast_queue_hangup(p->owner);
else
@@ -10816,6 +10796,18 @@
}
}
+/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
+static void stop_media_flows(struct sip_pvt *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 */
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
{
@@ -10831,35 +10823,26 @@
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
- if (p->rtp)
+ if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp);
- if (p->vrtp)
+ if (p->owner)
+ pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
+ }
+ if (p->vrtp) {
videoqos = ast_rtp_get_quality(p->vrtp);
+ if (p->owner)
+ pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
+ }
/* Get RTCP quality before end of call */
- if (global.recordhistory) {
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
if (p->rtp)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
}
- if (p->rtp) {
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
- /* Immediately stop RTP */
- ast_rtp_stop(p->rtp);
- }
- if (p->vrtp) {
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
- /* Immediately stop VRTP */
- ast_rtp_stop(p->vrtp);
- }
- if (p->udptl) {
- /* Immediately stop UDPTL */
- ast_udptl_stop(p->udptl);
- }
+ stop_media_flows(p);
if (!ast_strlen_zero(get_header(req, "Also"))) {
ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
ast_inet_ntoa(p->recv.sin_addr));
@@ -11965,7 +11948,7 @@
}
/*! \brief Find authentication for a specific realm */
-static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm)
+GNURK struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm)
{
struct sip_auth *a;
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=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3.h Fri Oct 20 10:31:27 2006
@@ -506,7 +506,7 @@
#define SIP_PROG_INBAND_NEVER (0 << 25)
#define SIP_PROG_INBAND_NO (1 << 25)
#define SIP_PROG_INBAND_YES (2 << 25)
-#define SIP_FREE_BIT (1 << 27) /*!< Undefined bit - not in use */
+#define SIP_NO_HISTORY (1 << 27) /*!< Supress recording request/response history */
#define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
#define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
#define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
@@ -921,5 +921,14 @@
extern struct expiry_times expiry; /*!< Various expiration times */
extern struct ast_config *notify_types; /*!< Notification types */
extern const char notify_config[]; /*!< Configuration file for notifications */
+extern const struct cfsip_methods sip_methods[]; /*!< sip3_parse.c: SIP method table */
+extern struct sip_globals global; /* Defined in chan_sip3.c */
+extern struct sched_context *sched; /*!< The scheduling context */
+extern struct io_context *io; /*!< The IO context */
+extern struct channel_counters sipcounters; /*!< Various object counters */
+extern struct sip_user_list userl; /*!< User list - will be gone soon ! */
+extern struct sip_device_list peerl; /*!< Device list */
+extern struct sip_register_list regl; /*!< Registration list */
+extern struct sip_auth *authl; /*!< Realm authentications */
#endif
Modified: team/oej/codename-pineapple/channels/sip3/sip3_auth.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_auth.c?rev=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_auth.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_auth.c Fri Oct 20 10:31:27 2006
@@ -279,7 +279,7 @@
/* No old challenge */
return -1;
}
- if (global.recordhistory)
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RegistryAuth", "Try: %d", p->authtries);
if (sip_debug_test_pvt(p) && p->registry)
ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
Modified: team/oej/codename-pineapple/channels/sip3/sip3_callerid.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_callerid.c?rev=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_callerid.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_callerid.c Fri Oct 20 10:31:27 2006
@@ -78,13 +78,14 @@
#include "asterisk/localtime.h"
#include "asterisk/compiler.h"
#include "sip3.h"
+#include "sip3funcs.h"
/*! \brief Get caller id number from Remote-Party-ID header field
* Returns true if number should be restricted (privacy setting found)
* output is set to NULL if no number found
*/
-static int get_rpid_num(const char *input, char *output, int maxlen)
+int get_rpid_num(const char *input, char *output, int maxlen)
{
char *start;
char *end;
@@ -112,7 +113,7 @@
}
/*! \brief Get caller id name from SIP headers */
-static char *get_calleridname(const char *input, char *output, size_t outputsize)
+char *get_calleridname(const char *input, char *output, size_t outputsize)
{
const char *end = strchr(input,'<'); /* first_bracket */
const char *tmp = strchr(input,'"'); /* first quote */
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=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_network.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_network.c Fri Oct 20 10:31:27 2006
@@ -87,7 +87,6 @@
#include "asterisk/abstract_jb.h"
#include "asterisk/compiler.h"
#include "sip3.h"
-#include "sip3core.h" /* Old stuff in the core file */
#include "sip3funcs.h" /* Moved functions */
struct sip_network sipnet; /* Socket and networking data */
@@ -190,7 +189,7 @@
}
p->recv = sin;
- if (global.recordhistory) /* This is a request or response, note what it was for */
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a request or response, note what it was for */
append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
if (!lockretry) {
@@ -435,7 +434,7 @@
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), req->data);
}
- if (global.recordhistory) {
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
@@ -461,7 +460,7 @@
else
ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
}
- if (global.recordhistory) {
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_method2txt(tmp.method));
Modified: team/oej/codename-pineapple/channels/sip3/sip3_parse.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_parse.c?rev=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_parse.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_parse.c Fri Oct 20 10:31:27 2006
@@ -90,7 +90,7 @@
#include "sip3.h"
/*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
-static const struct cfsip_methods sip_methods[] = {
+const struct cfsip_methods sip_methods[] = {
{ SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_NOT_CREATE_DIALOG},
{ SIP_RESPONSE, NO_RTP, "SIP/2.0" , CAN_NOT_CREATE_DIALOG},
{ SIP_REGISTER, NO_RTP, "REGISTER" , CAN_CREATE_DIALOG},
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=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_sdprtp.c Fri Oct 20 10:31:27 2006
@@ -746,7 +746,7 @@
}
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (chan->_state != AST_STATE_UP) { /* We are in early state */
- if (global.recordhistory)
+ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
if (option_debug)
ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip));
@@ -788,6 +788,8 @@
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
res = AST_RTP_TRY_NATIVE;
+ else if (ast_test_flag(&global.jbconf, AST_JB_FORCED))
+ res = AST_RTP_GET_FAILED;
ast_mutex_unlock(&p->lock);
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=45763&r1=45762&r2=45763&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3funcs.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3funcs.h Fri Oct 20 10:31:27 2006
@@ -43,14 +43,6 @@
/*! Chan_sip3.c */
/* XXX Should we really expose functions in chan_sip3.c ? */
-extern struct sip_globals global; /* Defined in chan_sip3.c */
-extern struct sched_context *sched; /*!< The scheduling context */
-extern struct io_context *io; /*!< The IO context */
-extern struct channel_counters sipcounters; /*!< Various object counters */
-extern struct sip_user_list userl;
-extern struct sip_device_list peerl;
-extern struct sip_register_list regl;
-extern struct sip_auth *authl;
GNURK inline int sip_debug_test_pvt(struct sip_pvt *p) ;
GNURK void append_history_full(struct sip_pvt *p, const char *fmt, ...);
@@ -70,6 +62,8 @@
GNURK void destroy_association(struct sip_peer *peer);
GNURK void reg_source_db(struct sip_peer *peer);
GNURK int expire_register(void *data);
+GNURK struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
+GNURK int add_line(struct sip_request *req, const char *line);
/*! sip3_refer.c */
GNURK const char *referstatus2str(enum referstatus rstatus) attribute_pure;
@@ -136,4 +130,8 @@
GNURK struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin);
GNURK struct sip_peer *realtime_user(const char *username);
GNURK int reload_config(enum channelreloadreason reason);
+
+/* sip3_callerid.c */
+GNURK int get_rpid_num(const char *input, char *output, int maxlen);
+GNURK char *get_calleridname(const char *input, char *output, size_t outputsize);
#endif
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