Sapian
/
asterisk
mirror of http://gerrit.asterisk.org/asterisk
3
0
Fork 0
Code Issues Releases Wiki Activity

various minor formatting changes and code cleanups (bug #4262)

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Kevin P. Fleming 20 years ago
parent e37adfa243
commit 4515e94c39
1 changed files with 66 additions and 34 deletions
Show Stats Download Patch File Download Diff File

100
channels/chan_sip.c
Unescape View File

@ -156,10 +156,10 @@ const struct cfsip_methods {
};
#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 @@ static int sip_indicate(struct ast_channel *ast, int condition)
}
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 @@ static int sip_indicate(struct ast_channel *ast, int condition)
/*--- 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 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
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 @@ static int sip_unescape_uri(char *uri)
/*--- 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 @@ static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_
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 @@ static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_
}
}
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 @@ static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_
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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
}
}
} 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 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
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);

Write Preview
Loading…
Cancel
Save