[Asterisk-cvs] asterisk/channels chan_sip.c,1.728,1.729
kpfleming at lists.digium.com
kpfleming at lists.digium.com
Sat May 14 20:21:47 CDT 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv32611/channels
Modified Files:
chan_sip.c
Log Message:
various minor formatting changes and code cleanups (bug #4262)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.728
retrieving revision 1.729
diff -u -d -r1.728 -r1.729
--- chan_sip.c 8 May 2005 16:49:28 -0000 1.728
+++ chan_sip.c 15 May 2005 00:27:18 -0000 1.729
@@ -156,10 +156,10 @@
};
-#define DEFAULT_SIP_PORT 5060 /* From RFC 2543 */
-#define SIP_MAX_PACKET 4096 /* Also from RFC 2543, should sub headers tho */
+#define DEFAULT_SIP_PORT 5060 /* From RFC 3261 (former 2543) */
+#define SIP_MAX_PACKET 4096 /* Also from RFC 3261 (2543), should sub headers tho */
-#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER"
+#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY"
static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;
@@ -2181,6 +2181,25 @@
}
res = -1;
break;
+ case AST_CONTROL_HOLD: /* We are put on hold */
+ /* The PBX is providing us with onhold music, but
+ should we clear the RTP stream with the other
+ end? Guess we could do that if there's no
+ musiconhold class defined for this channel
+ */
+ if (sipdebug)
+ ast_log(LOG_DEBUG, "SIP dialog on hold: %s\n", p->callid);
+ res = -1;
+ ast_set_flag(p, SIP_CALL_ONHOLD);
+ break;
+ case AST_CONTROL_UNHOLD: /* We are back from hold */
+ /* Open RTP stream if we decide to close it
+ */
+ if (sipdebug)
+ ast_log(LOG_DEBUG, "SIP dialog off hold: %s\n", p->callid);
+ res = -1;
+ ast_clear_flag(p, SIP_CALL_ONHOLD);
+ break;
case -1:
res = -1;
break;
@@ -2196,7 +2215,7 @@
/*--- sip_new: Initiate a call in the SIP channel */
-/* called from sip_request_call (calls from the pbx ) */
+/* called from sip_request (calls from the pbx ) */
static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
{
struct ast_channel *tmp;
@@ -3456,7 +3475,7 @@
else
c = p->initreq.rlPart2;
} else if (!ast_strlen_zero(p->okcontacturi)) {
- c = p->okcontacturi; /* Use for BYE or REINVITE */
+ c = p->okcontacturi; /* Use for BYE, REFER or REINVITE */
} else if (!ast_strlen_zero(p->uri)) {
c = p->uri;
} else {
@@ -5552,7 +5571,7 @@
-/*--- get_refer_info: Call transfer support (new standard) ---*/
+/*--- get_refer_info: Call transfer support (the REFER method) ---*/
static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_req)
{
@@ -5576,24 +5595,28 @@
refer_to = ditch_braces(h_refer_to);
if (!( (p_referred_by = get_header(req, "Referred-By")) && (h_referred_by = ast_strdupa(p_referred_by)) )) {
- ast_log(LOG_WARNING, "No Refer-To Header That's illegal\n");
- return -1;
+ ast_log(LOG_WARNING, "No Referrred-By Header That's not illegal\n");
+ } else {
+ referred_by = ditch_braces(h_referred_by);
}
-
- referred_by = ditch_braces(h_referred_by);
h_contact = get_header(req, "Contact");
- if (strncmp(refer_to, "sip:", 4) && strncmp(referred_by, "sip:", 4)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", refer_to);
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", referred_by);
+ if (strncmp(refer_to, "sip:", 4)) {
+ ast_log(LOG_WARNING, "Refer-to: Huh? Not a SIP header (%s)?\n", refer_to);
return -1;
}
+
+ if (strncmp(referred_by, "sip:", 4)) {
+ ast_log(LOG_WARNING, "Referred-by: Huh? Not a SIP header (%s) Ignoring?\n", referred_by);
+ referred_by = NULL;
+ }
+
refer_to += 4;
referred_by += 4;
if ((ptr = strchr(refer_to, '?'))) {
- /* Search for arguemnts */
+ /* Search for arguments */
*ptr = '\0';
ptr++;
if (!strncasecmp(ptr, "REPLACES=", 9)) {
@@ -5613,19 +5636,22 @@
}
}
- if ((ptr = strchr(refer_to, '@')))
+ if ((ptr = strchr(refer_to, '@'))) /* Skip domain (should be saved in SIPDOMAIN) */
*ptr = '\0';
if ((ptr = strchr(refer_to, ';')))
*ptr = '\0';
- if ((ptr = strchr(referred_by, '@')))
- *ptr = '\0';
- if ((ptr = strchr(referred_by, ';')))
- *ptr = '\0';
+ if (referred_by) {
+ if ((ptr = strchr(referred_by, '@')))
+ *ptr = '\0';
+ if ((ptr = strchr(referred_by, ';')))
+ *ptr = '\0';
+ }
if (sip_debug_test_pvt(sip_pvt)) {
- ast_verbose("Looking for %s in %s\n", refer_to, sip_pvt->context);
- ast_verbose("Looking for %s in %s\n", referred_by, sip_pvt->context);
+ ast_verbose("Transfer to %s in %s\n", refer_to, sip_pvt->context);
+ if (referred_by)
+ ast_verbose("Transfer from %s in %s\n", referred_by, sip_pvt->context);
}
if (!ast_strlen_zero(replace_callid)) {
/* This is a supervised transfer */
@@ -5646,15 +5672,15 @@
ast_log(LOG_NOTICE, "Supervised transfer requested, but unable to find callid '%s'. Both legs must reside on Asterisk box to transfer at this time.\n", replace_callid);
/* XXX The refer_to could contain a call on an entirely different machine, requiring an
INVITE with a replaces header -anthm XXX */
-
-
+ /* The only way to find out is to use the dialplan - oej */
}
} else if (ast_exists_extension(NULL, sip_pvt->context, refer_to, 1, NULL) || !strcmp(refer_to, ast_parking_ext())) {
- /* This is an unsupervised transfer */
+ /* This is an unsupervised transfer (blind transfer) */
- ast_log(LOG_DEBUG,"Assigning Extension %s to REFER-TO\n", refer_to);
- ast_log(LOG_DEBUG,"Assigning Extension %s to REFERRED-BY\n", referred_by);
- ast_log(LOG_DEBUG,"Assigning Contact Info %s to REFER_CONTACT\n", h_contact);
+ ast_log(LOG_DEBUG,"Unsupervised transfer to (Refer-To): %s\n", refer_to);
+ if (referred_by)
+ ast_log(LOG_DEBUG,"Transferred by (Referred-by: ) %s \n", referred_by);
+ ast_log(LOG_DEBUG,"Transfer Contact Info %s (REFER_CONTACT)\n", h_contact);
ast_copy_string(sip_pvt->refer_to, refer_to, sizeof(sip_pvt->refer_to));
ast_copy_string(sip_pvt->referred_by, referred_by, sizeof(sip_pvt->referred_by));
if (h_contact) {
@@ -8536,6 +8562,9 @@
int gotdest;
struct ast_frame af = { AST_FRAME_NULL, };
+ /* Check if this is a loop */
+ /* This happens since we do not properly support SIP domain
+ handling yet... -oej */
if (ast_test_flag(p, SIP_OUTGOING) && p->owner && (p->owner->_state != AST_STATE_UP)) {
/* This is a call to ourself. Send ourselves an error code and stop
processing immediately, as SIP really has no good mechanism for
@@ -8547,7 +8576,7 @@
if (!ignore) {
/* Use this as the basis */
if (debug)
- ast_verbose("Using latest request as basis request\n");
+ ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
sip_cancel_destroy(p);
/* This call is no longer outgoing if it ever was */
ast_clear_flag(p, SIP_OUTGOING);
@@ -8557,7 +8586,7 @@
check_via(p, req);
if (p->owner) {
/* Handle SDP here if we already have an owner */
- if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
+ if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
if (process_sdp(p, req)) {
transmit_response(p, "488 Not acceptable here", req);
ast_set_flag(p, SIP_NEEDDESTROY);
@@ -8602,11 +8631,11 @@
if (ast_strlen_zero(p->context))
strcpy(p->context, default_context);
/* Check number of concurrent calls -vs- incoming limit HERE */
- ast_log(LOG_DEBUG, "Check for res for %s\n", p->username);
- res = update_user_counter(p,INC_IN_USE);
+ ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username);
+ res = update_user_counter(p, INC_IN_USE);
if (res) {
if (res < 0) {
- ast_log(LOG_DEBUG, "Failed to place call for user %s, too many calls\n", p->username);
+ ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
if (ignore)
transmit_response(p, "480 Temporarily Unavailable (Call limit)", req);
else
@@ -8653,8 +8682,11 @@
}
}
- } else
+ } else {
+ if (option_debug > 1 && sipdebug)
+ ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid);
c = p->owner;
+ }
if (!ignore && p)
p->lastinvite = seqno;
if (c) {
@@ -8741,7 +8773,7 @@
struct ast_channel *transfer_to;
if (option_debug > 2)
- ast_log(LOG_DEBUG, "We found a REFER!\n");
+ ast_log(LOG_DEBUG, "SIP call transfer received for call %s (REFER)!\n", p->callid);
if (ast_strlen_zero(p->context))
strcpy(p->context, default_context);
res = get_refer_info(p, req);
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