[asterisk-commits] rizzo: branch rizzo/astobj2 r57984 -
/team/rizzo/astobj2/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Mar 6 01:48:07 MST 2007
Author: rizzo
Date: Tue Mar 6 02:48:06 2007
New Revision: 57984
URL: http://svn.digium.com/view/asterisk?view=rev&rev=57984
Log:
merge from trunk to svn 57478 (current as of today)
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=57984&r1=57983&r2=57984
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Tue Mar 6 02:48:06 2007
@@ -825,11 +825,12 @@
#define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< Call states */
#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (1 << 23) /*!< 23: One directional hold */
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (1 << 24) /*!< 24: Inactive */
-#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< 25: ???? */
+#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< 25: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< 26: Buggy CISCO MWI fix */
-#define SIP_PAGE2_NOTEXT (1 << 27) /*!< 26: Text not supported */
-#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< 27: Global text enable */
-#define SIP_PAGE2_DEBUG_TEXT (1 << 29) /*!< 28: Global text debug */
+#define SIP_PAGE2_NOTEXT (1 << 27) /*!< 27: Text not supported */
+#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< 28: Global text enable */
+#define SIP_PAGE2_DEBUG_TEXT (1 << 29) /*!< 29: Global text debug */
+#define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< 30: Is this an outgoing call? */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
@@ -3597,7 +3598,7 @@
{
char name[256];
int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
- int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
+ int outgoing = ast_test_flag(&fup->flags[1], SIP_PAGE2_OUTGOING_CALL);
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
@@ -3871,6 +3872,11 @@
}
if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
+ if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+ if (option_debug && sipdebug)
+ ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
+ update_call_counter(p, DEC_CALL_LIMIT);
+ }
if (option_debug >3)
ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); /* also cancels previous one if there */
@@ -3894,9 +3900,11 @@
ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");
sip_pvt_lock(p);
- if (option_debug && sipdebug)
- ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
- update_call_counter(p, DEC_CALL_LIMIT);
+ if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+ if (option_debug && sipdebug)
+ ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
+ update_call_counter(p, DEC_CALL_LIMIT);
+ }
/* Determine how to disconnect */
if (p->owner != ast) {
@@ -4702,27 +4710,33 @@
(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)) {
+ /* We already hold the channel lock */
+ if (!p->owner || f->frametype != AST_FRAME_VOICE)
+ return f;
+
+ if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
+ if (!(f->subclass & p->jointcapability)) {
+ if (option_debug) {
+ ast_log(LOG_DEBUG, "Bogus frame of format '%s' received from '%s'!\n",
+ ast_getformatname(f->subclass), p->owner->name);
+ }
+ return &ast_null_frame;
+ }
+ 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 | AST_FORMAT_TEXT_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, "Oooh, format changed to %d\n", f->subclass);
- p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_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);
- }
- }
+ 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);
}
}
}
@@ -8502,7 +8516,7 @@
if (contact)
ast_copy_string(peer->fullcontact, contact, sizeof(peer->fullcontact));
- if (option_verbose > 2)
+ if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_3 "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n",
peer->name, peer->username, ast_inet_ntoa(in), port, expiry);
@@ -13598,7 +13612,11 @@
} else if (sipmethod == SIP_NOTIFY) {
/* They got the notify, this is the end */
if (p->owner) {
- ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
+ if (p->refer) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Got 200 OK on NOTIFY for transfer\n");
+ } else
+ ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
/* ast_queue_hangup(p->owner); Disabled */
} else {
if (!p->subscribed && !p->refer)
@@ -13889,7 +13907,7 @@
ast_log(LOG_DEBUG, "-- No target second channel ---\n");
ast_log(LOG_DEBUG, "-- END Sip transfer:--------------------\n");
}
- if (transferer->chan2) { /* We have a bridge on the transferer's channel */
+ if (transferer->chan2) { /* We have a bridge on the transferer's channel */
peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */
peerb = target->chan1; /* Transferer - PBX -> target channel - This will get lost in masq */
peerc = transferer->chan2; /* Asterisk to Transferee */
@@ -15840,6 +15858,12 @@
}
return res;
}
+ }
+
+ if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY)) {
+ transmit_response(p, "503 Server error", req);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ return -1;
}
/* Handle various incoming SIP methods in requests */
@@ -16557,6 +16581,8 @@
*cause = AST_CAUSE_SWITCH_CONGESTION;
return NULL;
}
+
+ ast_set_flag(&p->flags[1], SIP_PAGE2_OUTGOING_CALL);
if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
sip_destroy(p);
More information about the asterisk-commits
mailing list