[svn-commits] oej: branch oej/codename-pineapple r47214 - in
/team/oej/codename-pineapple/c...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon Nov 6 00:18:31 MST 2006
Author: oej
Date: Mon Nov 6 01:18:30 2006
New Revision: 47214
URL: http://svn.digium.com/view/asterisk?rev=47214&view=rev
Log:
Update from Berlin
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
team/oej/codename-pineapple/channels/sip3/sip3.h
team/oej/codename-pineapple/channels/sip3/sip3_compose.c
team/oej/codename-pineapple/channels/sip3/sip3_dialog.c
team/oej/codename-pineapple/channels/sip3/sip3_parse.c
team/oej/codename-pineapple/channels/sip3/sip3_services.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=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Mon Nov 6 01:18:30 2006
@@ -298,7 +298,10 @@
* state engine works properly.
* \title Random thoughts
- * - Integrate sip_pkt and sip_request
+ * - Integrate sip_pkt and sip_request - done
+ * - Mark the UNACKed packets in dialog->packets with a flag
+ * Maybe two lists of packets, to keep it simple for do_monitor?
+ * Or a flag when we have no open transactions?
* - Use sip-request as "transaction holders" within sip_dialog
* - Keep them in Cseq order
* - Add transaction state
@@ -631,15 +634,56 @@
return sip_debug_test_addr(sip_real_dst(dialog));
}
+/*! \brief Find via branch parameter */
+static void find_via_branch(struct sip_dialog *dialog, struct sip_request *req)
+{
+ char *dupvia = ast_strdupa(req->via);
+ char *viabranch;
+ char *sep;
+
+ if (ast_strlen_zero(req->via))
+ return;
+ dupvia = ast_strdupa(req->via);
+ if (!(viabranch = strcasestr(dupvia, ";branch=")))
+ return;
+ viabranch += 8;
+ if ((sep = strchr(viabranch, ';')))
+ *sep = '\0';
+ if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug > 3)
+ ast_log(LOG_DEBUG, "* Found via branch %s\n", viabranch);
+ ast_string_field_set(dialog, remotebranch, viabranch);
+}
+
+
+/*! \brief Make branch tag for via header if it does not exist yet */
+static char *ourdialogbranch(struct sip_dialog *dialog, int forcenewbranch)
+{
+ char branch[20];
+ int seed = 0;
+
+ if (forcenewbranch || ast_strlen_zero(dialog->ourbranch)) {
+ if (forcenewbranch)
+ seed ^= ast_random();
+ else
+ seed = ast_random();
+ snprintf(branch, sizeof(branch), "z9hG4bk%08x", seed);
+ ast_string_field_set(dialog, ourbranch, branch);
+ }
+
+ return((char *) dialog->ourbranch);
+
+}
+
/*! \brief Build a Via header for a request */
-GNURK void build_via(struct sip_dialog *dialog)
+GNURK void build_via(struct sip_dialog *dialog, int forcenewbranch)
{
/* Work around buggy UNIDEN UIP200 firmware */
const char *rport = ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
/* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
- ast_string_field_build(dialog, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
- ast_inet_ntoa(dialog->ourip), sipnet_ourport(), dialog->branch, rport);
+ ast_string_field_build(dialog, via, "SIP/2.0/UDP %s:%d;branch=%s%s",
+ ast_inet_ntoa(dialog->ourip), sipnet_ourport(),
+ ourdialogbranch(dialog, forcenewbranch), rport);
}
GNURK void append_history_full(struct sip_dialog *p, const char *fmt, ...)
@@ -1933,8 +1977,12 @@
determine_firstline_parts(req);
/* Determine the seqno of this request once and for all */
+ req->callid = get_header(req, "Call-ID");
+ req->from = get_header(req, "From");
+ req->to = get_header(req, "To");
+ req->via = get_header(req, "Via"); /* Get the first via header only */
+ req->cseqheader = get_header(req, "CSeq");
/* Seqno can be zero, but anyway... */
- req->cseqheader = get_header(req, "CSeq");
if (!req->seqno && sscanf(req->cseqheader, "%d ", &seqno) != 1)
req->seqno = seqno;
}
@@ -2544,8 +2592,7 @@
req.method = sipmethod;
if (init) { /* Seems like init always is 2 */
/* Bump branch even on initial requests */
- p->branch ^= ast_random();
- build_via(p);
+ build_via(p, TRUE);
if (init > 1)
initreqprep(&req, p, sipmethod);
else
@@ -4685,7 +4732,7 @@
struct sip_request req;
struct ast_variable *var;
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) {
+ if (!(p = sip_alloc(NULL, NULL, FALSE, SIP_NOTIFY))) {
ast_log(LOG_WARNING, "Unable to build sip pvt data for notify (memory/socket error)\n");
return RESULT_FAILURE;
}
@@ -4705,7 +4752,7 @@
/* Recalculate our side, and recalculate Call ID */
if (sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
p->ourip = sipnet.__ourip;
- build_via(p);
+ build_via(p, FALSE);
build_callid_pvt(p);
ast_cli(fd, "Sending NOTIFY of type '%s' to '%s'\n", argv[2], argv[i]);
transmit_sip_request(p, &req);
@@ -5370,7 +5417,7 @@
if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
char tag[128];
- gettag(req, "To", tag, sizeof(tag));
+ gettag(req->to, tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
if (p->relatedpeer && p->method == SIP_OPTIONS) {
@@ -5620,7 +5667,7 @@
the final reply to our INVITE */
char tag[128];
- gettag(req, "To", tag, sizeof(tag));
+ gettag(req->to, tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
@@ -7091,7 +7138,7 @@
if (ast_strlen_zero(p->theirtag)) {
char tag[128];
- gettag(req, "From", tag, sizeof(tag));
+ gettag(req->from, tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);
@@ -7203,7 +7250,7 @@
p = peer->mwipvt;
} else {
/* Build temporary dialog for this message */
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY)))
+ if (!(p = sip_alloc(NULL, NULL, FALSE, SIP_NOTIFY)))
return -1;
if (create_addr_from_peer(p, peer)) {
/* Maybe they're not registered, etc. */
@@ -7213,7 +7260,7 @@
/* Recalculate our side, and recalculate Call ID */
if (sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
p->ourip = sipnet.__ourip;
- build_via(p);
+ build_via(p, FALSE);
build_callid_pvt(p);
/* Destroy this session after 32 secs */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -7473,7 +7520,7 @@
ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\n");
sip_destroy(peer->call);
}
- if (!(p = peer->call = sip_alloc(NULL, NULL, 0, SIP_OPTIONS)))
+ if (!(p = peer->call = sip_alloc(NULL, NULL, FALSE, SIP_OPTIONS)))
return -1;
p->sa = peer->addr;
@@ -7493,7 +7540,7 @@
/* Recalculate our side, and recalculate Call ID */
if (sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
p->ourip = sipnet.__ourip;
- build_via(p);
+ build_via(p, FALSE);
build_callid_pvt(p);
if (peer->pokeexpire > -1)
@@ -7609,7 +7656,7 @@
if (option_debug)
ast_log(LOG_DEBUG, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
+ if (!(p = sip_alloc(NULL, NULL, FALSE, SIP_INVITE))) {
ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data);
*cause = AST_CAUSE_SWITCH_CONGESTION;
return NULL;
@@ -7646,7 +7693,7 @@
/* Recalculate our side, and recalculate Call ID */
if (sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
p->ourip = sipnet.__ourip;
- build_via(p);
+ build_via(p, FALSE);
build_callid_pvt(p);
/* We have an extension to call, don't use the full contact here */
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=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3.h Mon Nov 6 01:18:30 2006
@@ -373,7 +373,11 @@
enum transaction_state state; /*!< State of this transaction */
char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
char *rlPart2; /*!< The Request URI or Response Status */
- const char *cseqheader; /*! Pointer to the cseq header */
+ const char *cseqheader; /*!< Pointer to the cseq header */
+ const char *callid; /*!< Pointer to the call ID header */
+ const char *via; /*!< Pointer to the Via header */
+ const char *to; /*!< Pointer to the To header */
+ const char *from; /*!< Pointer to the from header */
int len; /*!< Length */
int headers; /*!< # of SIP Headers */
unsigned int flags; /*!< SIP_PKT Flags for this packet */
@@ -711,7 +715,8 @@
AST_STRING_FIELD(callid); /*!< Dialog ID: Global CallID - the call ID is a unique ID for this SIP dialog,
a string that never changes during the dialog */
AST_STRING_FIELD(theirtag); /*!< Dialog ID: remote side's tag */
- AST_STRING_FIELD(thisbranch); /*!< Dialog ID: branch tag from topmost via */
+ AST_STRING_FIELD(remotebranch); /*!< Dialog ID: branch tag from topmost via */
+ AST_STRING_FIELD(ourbranch); /*!< Dialog ID: branch tag from topmost via */
AST_STRING_FIELD(randdata); /*!< Random data */
AST_STRING_FIELD(realm); /*!< Authorization realm for this dialog */
AST_STRING_FIELD(nonce); /*!< Authorization nonce */
@@ -759,7 +764,6 @@
unsigned int sipoptions; /*!< Supported SIP options on the other end */
int authtries; /*!< Times we've tried to authenticate */
int expiry; /*!< How long we take to expire */
- long branch; /*!< The branch identifier of this session */
char tag[11]; /*!< Dialog ID: Our tag for this session */
int callingpres; /*!< Caller ID presentation settings */
struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
Modified: team/oej/codename-pineapple/channels/sip3/sip3_compose.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_compose.c?rev=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_compose.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_compose.c Mon Nov 6 01:18:30 2006
@@ -133,7 +133,7 @@
}
/*! \brief Prepare SIP response packet */
-int respprep(struct sip_request *resp, struct sip_dialog *p, const char *msg, const struct sip_request *req)
+int respprep(struct sip_request *resp, struct sip_dialog *p, const char *msg, struct sip_request *req)
{
char newto[256];
const char *ot;
@@ -142,8 +142,11 @@
copy_via_headers(p, resp, req, "Via");
if (msg[0] == '2')
copy_all_header(resp, req, "Record-Route");
- copy_header(resp, req, "From");
- ot = get_header(req, "To");
+ copy_header(resp, req, "From"); /* XXX this can be simplified when we are sure that req->from works*/
+ if (ast_strlen_zero(req->to))
+ req->to = get_header(req, "To");
+ ot = req->to;
+
if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) {
/* Add the proper tag if we don't have it already. If they have specified
their tag, use it. Otherwise, use our own tag */
@@ -156,8 +159,8 @@
ot = newto;
}
add_header(resp, "To", ot);
- copy_header(resp, req, "Call-ID");
- copy_header(resp, req, "CSeq");
+ copy_header(resp, req, "Call-ID"); /* Should use req->callid */
+ copy_header(resp, req, "CSeq"); /* Should use req->cseqheader */
add_header(resp, "User-Agent", global.useragent);
add_header(resp, "Allow", ALLOWED_METHODS);
add_header(resp, "Supported", SUPPORTED_EXTENSIONS);
@@ -173,7 +176,7 @@
snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry);
add_header(resp, "Contact", contact); /* Not when we unregister */
}
- } else if (msg[0] != '4' && p->our_contact[0]) {
+ } else if (msg[0] != '4' && p->our_contact[0] && req->method != SIP_BYE && req->method != SIP_CANCEL) {
add_header(resp, "Contact", p->our_contact);
}
return 0;
@@ -324,10 +327,7 @@
seqno = p->ocseq;
}
- if (newbranch) {
- p->branch ^= ast_random();
- build_via(p);
- }
+ build_via(p, newbranch);
/* Check for strict or loose router */
if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop,";lr") == NULL) {
Modified: team/oej/codename-pineapple/channels/sip3/sip3_dialog.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_dialog.c?rev=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_dialog.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_dialog.c Mon Nov 6 01:18:30 2006
@@ -92,7 +92,7 @@
#include "sip3funcs.h"
/*! \page chan_sip3_dialogs Chan_sip3: The dialog list
- \par The dialog list
+ \title The dialog list
The dialog list contains all active dialogs in various states.
A dialog can be
- an active call
@@ -100,7 +100,9 @@
- a subscription
- an inbound or outbound registration
- We will implement dialog states soon
+ \title Dialog states
+ Dialog states affect operation, especially in an INVITE
+ dialog. We now try to change dialog state in a clear way
\ref enum dialogstate
*/
@@ -120,12 +122,16 @@
void dialoglist_lock(void)
{
ast_mutex_lock(&dialoglock);
+ if (option_debug > 3)
+ ast_log(LOG_DEBUG, "=== SIP dialog list: LOCKED\n");
}
/*! \brief Unlock list of active SIP dialogs */
void dialoglist_unlock(void)
{
ast_mutex_unlock(&dialoglock);
+ if (option_debug > 3)
+ ast_log(LOG_DEBUG, "=== SIP dialog list: UNLOCKED\n");
}
/*! \brief Convert SIP dialog states to string */
@@ -161,14 +167,23 @@
/*! \brief Change dialog state for a SIP dialog and output to debug */
void dialogstatechange(struct sip_dialog *dialog, enum dialogstate newstate)
{
- dialog->state = newstate;
- if (sipdebug && option_debug > 1)
- ast_log(LOG_DEBUG, "-- Dialog %s changed state to %s\n", dialog->callid, dialogstate2str(newstate));
+ if (dialog->state == newstate) {
+ if (option_debug > 3)
+ ast_log(LOG_DEBUG, "Asked to change state to dialog that already has requested state: %s State %s\n", dialog->callid, dialogstate2str(newstate));
+ } else {
+ dialog->state = newstate;
+ if (sipdebug && option_debug > 1)
+ ast_log(LOG_DEBUG, "-- Dialog %s changed state to %s\n", dialog->callid, dialogstate2str(newstate));
+ if (global.recordhistory)
+ append_history(dialog, "DialogState", "New state: %s O-Cseq %d I-Cseq %d", dialogstate2str(newstate), dialog->ocseq, dialog->icseq);
+ }
}
/*! \brief For a reliable transmission, we need to get an reply to stop retransmission.
- Acknowledges receipt of a packet and stops retransmission */
+ Acknowledges receipt of a packet and stops retransmission
+ \note Assume that the dialog is locked.
+ */
/* We need a method for responses too ... */
void __sip_ack(struct sip_dialog *dialog, int seqno, int resp, int sipmethod, int reset)
{
@@ -226,6 +241,7 @@
}
/*! \brief Acks receipt of packet, keep it around (used for provisional responses)
+ \note Assume that the dialog is locked.
*/
int __sip_semi_ack(struct sip_dialog *dialog, int seqno, int resp, int sipmethod)
{
@@ -630,7 +646,6 @@
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;
@@ -678,7 +693,7 @@
if (p->method != SIP_REGISTER)
ast_string_field_set(p, fromdomain, global.default_fromdomain);
- build_via(p);
+ build_via(p, TRUE);
if (!callid) /* Make sure we have a unique call ID */
build_callid_pvt(p);
else
@@ -767,7 +782,6 @@
} else
p->ourip = sipnet.__ourip;
- p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
p->ocseq = INITIAL_CSEQ;
@@ -778,7 +792,7 @@
}
ast_string_field_set(p, fromdomain, global.default_fromdomain);
- build_via(p);
+ build_via(p, TRUE);
ast_string_field_set(p, callid, callid);
/* Use this temporary pvt structure to send the message */
@@ -794,88 +808,98 @@
Called by handle_request, sipsock_read */
struct sip_dialog *match_or_create_dialog(struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
{
- struct sip_dialog *p = NULL;
+ struct sip_dialog *cur = NULL;
char *tag = ""; /* note, tag is never NULL */
char totag[128];
char fromtag[128];
- const char *callid = get_header(req, "Call-ID");
- const char *from = get_header(req, "From");
- const char *to = get_header(req, "To");
- const char *cseq = get_header(req, "Cseq");
+
+ if (ast_strlen_zero(req->callid))
+ req->callid = get_header(req, "Call-ID");
+ if (ast_strlen_zero(req->from))
+ req->from = get_header(req, "From");
+ if (ast_strlen_zero(req->to))
+ req->to = get_header(req, "To");
+ if (ast_strlen_zero(req->cseqheader))
+ req->cseqheader = get_header(req, "Cseq");
/* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
/* get_header always returns non-NULL so we must use ast_strlen_zero() */
- if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
- ast_strlen_zero(from) || ast_strlen_zero(cseq))
+ if (ast_strlen_zero(req->callid) || ast_strlen_zero(req->to) ||
+ ast_strlen_zero(req->from) || ast_strlen_zero(req->cseqheader))
return NULL; /* Invalid packet */
/* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
we need more to identify a branch - so we have to check branch, from
and to tags to identify a call leg.
*/
- if (gettag(req, "To", totag, sizeof(totag)))
+ if (gettag(req->to, totag, sizeof(totag)))
ast_set_flag(req, SIP_PKT_WITH_TOTAG); /* Used in handle_request/response */
- gettag(req, "From", fromtag, sizeof(fromtag));
+ gettag(req->from, fromtag, sizeof(fromtag));
tag = (req->method == SIP_RESPONSE) ? totag : fromtag;
if (option_debug > 4 )
- ast_log(LOG_DEBUG, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);
+ ast_log(LOG_DEBUG, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", req->callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);
dialoglist_lock();
- for (p = dialoglist; p; p = p->next) {
+ for (cur = dialoglist; cur; cur = cur->next) {
/* we do not want packets with bad syntax to be connected to a PVT */
int found = FALSE;
if (req->method == SIP_REGISTER)
- found = (!strcmp(p->callid, callid));
+ found = (!strcmp(cur->callid, req->callid));
else
- found = (!strcmp(p->callid, callid) &&
- (!tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ;
+ found = (!strcmp(cur->callid, req->callid) &&
+ (!tag || ast_strlen_zero(cur->theirtag) || !strcmp(cur->theirtag, tag))) ;
if (option_debug > 4)
- ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);
+ ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", cur->callid, cur->theirtag, cur->tag);
/* If we get a new request within an existing to-tag - check the to tag as well */
if (found && req->method != SIP_RESPONSE) { /* SIP Request */
- if (p->tag[0] == '\0' && totag[0]) {
+ if (cur->tag[0] == '\0' && totag[0]) {
/* We have no to tag, but they have. Wrong dialog */
found = FALSE;
} else if (totag[0]) { /* Both have tags, compare them */
- if (strcmp(totag, p->tag)) {
+ if (strcmp(totag, cur->tag)) {
found = FALSE; /* This is not our packet */
}
}
if (!found && option_debug > 4)
- ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_method2txt(req->method));
+ ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", cur->callid, totag, sip_method2txt(req->method));
}
if (found) {
/* Found the call */
- ast_mutex_lock(&p->lock);
+ ast_mutex_lock(&cur->lock);
dialoglist_unlock();
- return p;
+ return cur;
}
}
dialoglist_unlock();
- if (sip_methods[intended_method].creates_dialog == CAN_CREATE_DIALOG) {
- 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))) {
+ if (sip_methods[intended_method].creates_dialog != CAN_CREATE_DIALOG && intended_method != SIP_RESPONSE) {
+ transmit_response_using_temp(req->callid, sin, TRUE, intended_method, req, "481 Call leg/transaction does not exist");
+ return cur;
+ }
+ switch (intended_method) {
+ case SIP_REFER:
+ /* We do not support out-of-dialog REFERs yet */
+ transmit_response_using_temp(req->callid, sin, TRUE, intended_method, req, "603 Declined (no dialog)");
+ break;
+ case SIP_NOTIFY:
+ /* We do not support out-of-dialog NOTIFY either,
+ like voicemail notification, so cancel that early */
+ transmit_response_using_temp(req->callid, sin, TRUE, intended_method, req, "489 Bad event");
+ break;
+ default:
+ /* ready to create a new dialog. */
+ if ((cur = sip_alloc(req->callid, sin, TRUE, intended_method))) {
/* This method creates dialog */
- /* Ok, we've created a dialog, let's go and process it */
- ast_mutex_lock(&p->lock);
+ /* Ok, we've created a dialog, let's go and process it */
+ ast_mutex_lock(&cur->lock);
}
- } else {
- if (intended_method != SIP_RESPONSE)
- transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist");
- }
-
- return p;
-}
+ break;
+ }
+
+ return cur;
+}
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=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_parse.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_parse.c Mon Nov 6 01:18:30 2006
@@ -196,15 +196,14 @@
* \return Returns the pointer to the provided tag buffer,
* or NULL if the tag was not found.
*/
-const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
+const char *gettag(const char *header, char *tagbuf, int tagbufsize)
{
const char *thetag;
if (!tagbuf)
return NULL;
tagbuf[0] = '\0'; /* reset the buffer */
- thetag = get_header(req, header);
- thetag = strcasestr(thetag, ";tag=");
+ thetag = strcasestr(header, ";tag=");
if (thetag) {
thetag += 5;
ast_copy_string(tagbuf, thetag, tagbufsize);
Modified: 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=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_services.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_services.c Mon Nov 6 01:18:30 2006
@@ -498,15 +498,12 @@
snprintf(addr, sizeof(addr), "sip:%s", S_OR(p->fromdomain, r->hostname));
ast_string_field_set(p, uri, addr);
- p->branch ^= ast_random();
-
init_req(&req, sipmethod, addr);
/* Add to CSEQ */
snprintf(tmp, sizeof(tmp), "%u %s", ++r->ocseq, sip_method2txt(sipmethod));
p->ocseq = r->ocseq;
-
- build_via(p);
+ build_via(p, TRUE);
add_header(&req, "Via", p->via);
add_header(&req, "From", from);
add_header(&req, "To", to);
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=47214&r1=47213&r2=47214&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3funcs.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3funcs.h Mon Nov 6 01:18:30 2006
@@ -69,7 +69,7 @@
GNURK int transmit_state_notify(struct sip_dialog *p, int state, int full, int timeout);
GNURK int transmit_request_with_auth(struct sip_dialog *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
GNURK void do_setnat(struct sip_dialog *p, int natflags);
-GNURK void build_via(struct sip_dialog *p);
+GNURK void build_via(struct sip_dialog *p, int forcenewbranch);
GNURK void __sip_pretend_ack(struct sip_dialog *p);
GNURK int create_addr(struct sip_dialog *dialog, const char *opeer);
GNURK void build_contact(struct sip_dialog *p);
@@ -127,7 +127,7 @@
GNURK const char *__get_header(const struct sip_request *req, const char *name, int *start);
GNURK char *get_in_brackets(char *tmp);
GNURK char *generate_random_string(char *buf, size_t size);
-GNURK const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
+GNURK const char *gettag(const char *header, char *tagbuf, int tagbufsize);
GNURK int determine_firstline_parts(struct sip_request *req);
GNURK void extract_uri(struct sip_dialog *p, struct sip_request *req);
@@ -136,7 +136,7 @@
GNURK void append_date(struct sip_request *req);
GNURK int add_text(struct sip_request *req, const char *text);
GNURK int add_digit(struct sip_request *req, char digit);
-GNURK int respprep(struct sip_request *resp, struct sip_dialog *p, const char *msg, const struct sip_request *req);
+GNURK int respprep(struct sip_request *resp, struct sip_dialog *p, const char *msg, struct sip_request *req);
GNURK void add_route(struct sip_request *req, struct sip_route *route);
GNURK int add_line(struct sip_request *req, const char *line);
GNURK int add_header_contentLength(struct sip_request *req, int len);
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