[asterisk-commits] branch oej/siptransfer r12547 - in
/team/oej/siptransfer: ./ apps/ cdr/ chann...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sun Mar 12 09:57:01 MST 2006
Author: oej
Date: Sun Mar 12 10:56:47 2006
New Revision: 12547
URL: http://svn.digium.com/view/asterisk?rev=12547&view=rev
Log:
Resolve conflict
Modified:
team/oej/siptransfer/ (props changed)
team/oej/siptransfer/README
team/oej/siptransfer/README.siptransfer
team/oej/siptransfer/UPGRADE.txt
team/oej/siptransfer/apps/app_queue.c
team/oej/siptransfer/ast_expr2.fl
team/oej/siptransfer/ast_expr2f.c
team/oej/siptransfer/cdr/cdr_tds.c
team/oej/siptransfer/channels/chan_iax2.c
team/oej/siptransfer/channels/chan_misdn.c
team/oej/siptransfer/channels/chan_sip.c
team/oej/siptransfer/channels/misdn/chan_misdn_config.h
team/oej/siptransfer/channels/misdn/isdn_lib.c
team/oej/siptransfer/channels/misdn/isdn_lib.h
team/oej/siptransfer/channels/misdn/isdn_msg_parser.c
team/oej/siptransfer/channels/misdn_config.c
team/oej/siptransfer/configs/cdr_tds.conf.sample
team/oej/siptransfer/configs/misdn.conf.sample
team/oej/siptransfer/doc/asterisk-mib.txt
team/oej/siptransfer/doc/misdn.txt
team/oej/siptransfer/include/asterisk/doxyref.h
team/oej/siptransfer/include/asterisk/sched.h
team/oej/siptransfer/sched.c
Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Mar 12 10:56:47 2006
@@ -1,1 +1,1 @@
-/trunk:1-12449
+/trunk:1-12544
Modified: team/oej/siptransfer/README
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/README?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/README (original)
+++ team/oej/siptransfer/README Sun Mar 12 10:56:47 2006
@@ -51,13 +51,15 @@
* All Wildcard (tm) products from Digium (www.digium.com)
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
* any full duplex sound card supported by ALSA or OSS
+ * any ISDN card supported by mISDN on Linux (BRI)
+ * The Xorcom AstriBank channel bank
* VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources.
-Check the voip-info.org wiki for more information on chan_capi, chan_misdn and
+Check the voip-info.org wiki for more information on chan_capi and
zaphfc.
-* UPGRADING FROM VERSION 1.0
+* UPGRADING FROM AN EARLIER VERSION
If you are updating from a previous version of Asterisk, make sure you
read the UPGRADE.txt file in the source directory. There are some files
Modified: team/oej/siptransfer/README.siptransfer
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/README.siptransfer?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/README.siptransfer (original)
+++ team/oej/siptransfer/README.siptransfer Sun Mar 12 10:56:47 2006
@@ -1,5 +1,8 @@
This is the SIP transfer re-write that now support attended transfers on multiple servers.
It significantly changes the way we handle incoming REFER transactions.
+
+Todo:
+- Resolve all the conflicts in the code properly
Additional features
- Improved transfer support (we can deny transfers now)
Modified: team/oej/siptransfer/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/UPGRADE.txt?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/UPGRADE.txt (original)
+++ team/oej/siptransfer/UPGRADE.txt Sun Mar 12 10:56:47 2006
@@ -59,7 +59,7 @@
Installation:
-* On BSD systems, the installation directories have changed to more "FreeBSDish" directories. On startup, Asterisk will look for the main configuration in /usr7local/etc/asterisk/asterisk.conf
+* On BSD systems, the installation directories have changed to more "FreeBSDish" directories. On startup, Asterisk will look for the main configuration in /usr/local/etc/asterisk/asterisk.conf
If you have an old installation, you might want to remove the binaries and move the configuration files to the new locations. The following directories are now default:
ASTLIBDIR /usr/local/lib/asterisk
ASTVARLIBDIR /usr/local/share/asterisk
Modified: team/oej/siptransfer/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_queue.c?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_queue.c (original)
+++ team/oej/siptransfer/apps/app_queue.c Sun Mar 12 10:56:47 2006
@@ -1624,6 +1624,14 @@
static void record_abandoned(struct queue_ent *qe)
{
ast_mutex_lock(&qe->parent->lock);
+ manager_event(EVENT_FLAG_AGENT, "QueueCallerAbandon",
+ "Queue: %s\r\n"
+ "Uniqueid: %s\r\n"
+ "Position: %d\r\n"
+ "OriginalPosition: %d\r\n"
+ "HoldTime: %d\r\n",
+ qe->parent->name, qe->chan->uniqueid, qe->pos, qe->opos, (int)(time(NULL) - qe->start));
+
qe->parent->callsabandoned++;
ast_mutex_unlock(&qe->parent->lock);
}
Modified: team/oej/siptransfer/ast_expr2.fl
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/ast_expr2.fl?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/ast_expr2.fl (original)
+++ team/oej/siptransfer/ast_expr2.fl Sun Mar 12 10:56:47 2006
@@ -201,7 +201,7 @@
s, parseio->string,spacebuf2);
#endif
#ifndef STANDALONE
- ast_log(LOG_WARNING,"If you have questions, please refer to doc/README.variables in the asterisk source.\n");
+ ast_log(LOG_WARNING,"If you have questions, please refer to doc/channelvariables.txt in the asterisk source.\n");
#endif
return(0);
}
Modified: team/oej/siptransfer/ast_expr2f.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/ast_expr2f.c?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/ast_expr2f.c (original)
+++ team/oej/siptransfer/ast_expr2f.c Sun Mar 12 10:56:47 2006
@@ -2981,7 +2981,7 @@
s, parseio->string,spacebuf2);
#endif
#ifndef STANDALONE
- ast_log(LOG_WARNING,"If you have questions, please refer to doc/README.variables in the asterisk source.\n");
+ ast_log(LOG_WARNING,"If you have questions, please refer to doc/channelvariables.txt in the asterisk source.\n");
#endif
return(0);
}
Modified: team/oej/siptransfer/cdr/cdr_tds.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/cdr/cdr_tds.c?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/cdr/cdr_tds.c (original)
+++ team/oej/siptransfer/cdr/cdr_tds.c Sun Mar 12 10:56:47 2006
@@ -89,6 +89,7 @@
static char *config = "cdr_tds.conf";
static char *hostname = NULL, *dbname = NULL, *dbuser = NULL, *password = NULL, *charset = NULL, *language = NULL;
+static char *table = NULL;
static int connected = 0;
@@ -135,7 +136,7 @@
sprintf(
sqlcmd,
- "INSERT INTO cdr "
+ "INSERT INTO %s "
"("
"accountcode, "
"src, "
@@ -175,6 +176,7 @@
"'%s', " /* amaflags */
"'%s'" /* uniqueid */
")",
+ table,
accountcode,
src,
dst,
@@ -415,6 +417,7 @@
if (password) free(password);
if (charset) free(charset);
if (language) free(language);
+ if (table) free(table);
return 0;
}
@@ -475,6 +478,13 @@
else
language = strdup("us_english");
+ ptr = ast_variable_retrieve(cfg,"global","table");
+ if (ptr == NULL) {
+ ast_log(LOG_DEBUG,"cdr_tds: table not specified. Assuming cdr\n");
+ ptr = "cdr";
+ }
+ table = strdup(ptr);
+
ast_config_destroy(cfg);
mssql_connect();
Modified: team/oej/siptransfer/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_iax2.c?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_iax2.c (original)
+++ team/oej/siptransfer/channels/chan_iax2.c Sun Mar 12 10:56:47 2006
@@ -5759,8 +5759,6 @@
ast_log(LOG_DEBUG, "Expiring registration for peer '%s'\n", p->name);
/* Reset the address */
memset(&p->addr, 0, sizeof(p->addr));
- /* Reset expire notice */
- p->expire = -1;
/* Reset expiry value */
p->expiry = min_reg_expire;
if (!ast_test_flag(p, IAX_TEMPONLY))
@@ -5778,6 +5776,9 @@
static int expire_registry(void *data)
{
+ struct iax2_peer *p = data;
+ /* Reset expire notice */
+ p->expire = -1;
#ifdef SCHED_MULTITHREADED
if (schedule_action(__expire_registry, data))
#endif
Modified: team/oej/siptransfer/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_misdn.c?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_misdn.c (original)
+++ team/oej/siptransfer/channels/chan_misdn.c Sun Mar 12 10:56:47 2006
@@ -648,13 +648,14 @@
{
struct ast_channel *ast=help->ast;
ast_cli(fd,
- "* Pid:%d Prt:%d Ch:%d Mode:%s Org:%s dad:%s oad:%s ctx:%s state:%s\n",
+ "* Pid:%d Prt:%d Ch:%d Mode:%s Org:%s dad:%s oad:%s rad:%s ctx:%s state:%s\n",
bc->pid, bc->port, bc->channel,
bc->nt?"NT":"TE",
help->orginator == ORG_AST?"*":"I",
ast?ast->exten:NULL,
ast?AST_CID_P(ast):NULL,
+ bc->rad,
ast?ast->context:NULL,
misdn_get_ch_state(help)
);
@@ -1340,26 +1341,47 @@
misdn_cfg_get( port, MISDN_CFG_LOCALDIALPLAN, &bc->onumplan, sizeof(int));
switch (bc->onumplan) {
case NUMPLAN_INTERNATIONAL:
- chan_misdn_log(2, port, " --> TON: International\n");
+ chan_misdn_log(2, port, " --> LTON: International\n");
break;
case NUMPLAN_NATIONAL:
- chan_misdn_log(2, port, " --> TON: National\n");
+ chan_misdn_log(2, port, " --> LTON: National\n");
break;
case NUMPLAN_SUBSCRIBER:
- chan_misdn_log(2, port, " --> TON: Subscriber\n");
+ chan_misdn_log(2, port, " --> LTON: Subscriber\n");
break;
case NUMPLAN_UNKNOWN:
- chan_misdn_log(2, port, " --> TON: Unknown\n");
+ chan_misdn_log(2, port, " --> LTON: Unknown\n");
break;
/* Maybe we should cut off the prefix if present ? */
default:
chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
break;
}
- }
-
-
-
+
+ misdn_cfg_get( port, MISDN_CFG_CPNDIALPLAN, &bc->cpnnumplan, sizeof(int));
+
+ switch (bc->cpnnumplan) {
+ case NUMPLAN_INTERNATIONAL:
+ chan_misdn_log(2, port, " --> CTON: International\n");
+ break;
+ case NUMPLAN_NATIONAL:
+ chan_misdn_log(2, port, " --> CTON: National\n");
+ break;
+ case NUMPLAN_SUBSCRIBER:
+ chan_misdn_log(2, port, " --> CTON: Subscriber\n");
+ break;
+ case NUMPLAN_UNKNOWN:
+ chan_misdn_log(2, port, " --> CTON: Unknown\n");
+ break;
+ /* Maybe we should cut off the prefix if present ? */
+ default:
+ chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
+ break;
+ }
+
+ }
+
+
} else { /** ORIGINATOR MISDN **/
Modified: team/oej/siptransfer/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_sip.c?rev=12547&r1=12546&r2=12547&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_sip.c (original)
+++ team/oej/siptransfer/channels/chan_sip.c Sun Mar 12 10:56:47 2006
@@ -103,6 +103,7 @@
#ifndef TRUE
#define TRUE 1
#endif
+
#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
#ifndef IPTOS_MINCOST
@@ -177,6 +178,13 @@
submitting a patch. If these two lists do not match properly
bad things will happen.
*/
+
+enum xmittype {
+ XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
+ If it fails, it's critical and will cause a teardown of the session */
+ XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
+ XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
+};
enum subscriptiontype {
NONE = 0,
@@ -977,11 +985,11 @@
/*---------------------------- Forward declarations of functions in chan_sip.c */
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req);
-static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
+static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable);
static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported);
-static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *rand, int reliable, const char *header, int stale);
-static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
-static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
+static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
+static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
+static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sendsdp, int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p);
static int transmit_info_with_digit(struct sip_pvt *p, char digit);
@@ -1015,7 +1023,7 @@
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */
static int check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
const char *secret, const char *md5secret, int sipmethod,
- char *uri, int reliable, int ignore);
+ char *uri, enum xmittype reliable, int ignore);
static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
static void append_date(struct sip_request *req); /* Append date to SIP packet */
static int determine_firstline_parts(struct sip_request *req);
@@ -1029,7 +1037,7 @@
static void parse_request(struct sip_request *req);
static char *get_header(struct sip_request *req, const char *name);
static void copy_request(struct sip_request *dst,struct sip_request *src);
-static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal);
+static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req);
static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader);
static int sip_poke_peer(struct sip_peer *peer);
static int __sip_do_register(struct sip_registry *r);
@@ -1466,7 +1474,9 @@
}
/* Too many retries */
if (pkt->owner && pkt->method != SIP_OPTIONS) {
- if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */ ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request"); } else {
+ if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
+ ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
+ } else {
if ((pkt->method == SIP_OPTIONS) && sipdebug)
ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
}
@@ -1536,6 +1546,7 @@
pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
if (fatal)
ast_set_flag(pkt, FLAG_FATAL);
+
if (pkt->timer_t1)
siptimer_a = pkt->timer_t1 * 2;
@@ -1721,7 +1732,7 @@
}
/*! \brief Transmit response on SIP request*/
-static int send_response(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
+static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
int res;
@@ -1738,7 +1749,7 @@
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
}
res = (reliable) ?
- __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1), req->method) :
+ __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
__sip_xmit(p, req->data, req->len);
if (res > 0)
return 0;
@@ -1746,7 +1757,7 @@
}
/*! \brief Send SIP Request to the other part of the dialogue */
-static int send_request(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
+static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
int res;
@@ -2816,13 +2827,16 @@
if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
if (needcancel) { /* Outgoing call, not up */
if (ast_test_flag(p, SIP_OUTGOING)) {
- transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
+ /* stop retransmitting an INVITE that has not received a response */
+ __sip_pretend_ack(p);
+
+ /* Send a new request: CANCEL */
+ transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, 0);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
ast_clear_flag(&locflags, SIP_NEEDDESTROY);
- sip_scheddestroy(p, 15000);
- /* stop retransmitting an INVITE that has not received a response */
- __sip_pretend_ack(p);
+
+ sip_scheddestroy(p, 32000);
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
only if the channel is not auto-congested */
@@ -2831,14 +2845,14 @@
} else { /* Incoming call, not up */
char *res;
if (ast->hangupcause && ((res = hangup_cause2sip(ast->hangupcause)))) {
- transmit_response_reliable(p, res, &p->initreq, 1);
+ transmit_response_reliable(p, res, &p->initreq);
} else
- transmit_response_reliable(p, "603 Declined", &p->initreq, 1);
+ transmit_response_reliable(p, "603 Declined", &p->initreq);
}
} else { /* Call is in UP state, send BYE */
if (!p->pendinginvite) {
/* Send a hangup */
- transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@@ -2853,12 +2867,34 @@
return 0;
}
+/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
+static void try_suggested_sip_codec(struct sip_pvt *p)
+{
+ int fmt;
+ const char *codec;
+
+ codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+ if (!codec)
+ return;
+
+ fmt = ast_getformatbyname(codec);
+ if (fmt) {
+ ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n", codec);
+ if (p->jointcapability & fmt) {
+ p->jointcapability &= fmt;
+ p->capability &= fmt;
+ } else
+ ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+ } else
+ ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
+ return;
+}
+
/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite
* Part of PBX interface */
static int sip_answer(struct ast_channel *ast)
{
- int res = 0,fmt;
- const char *codec;
+ int res = 0;
struct sip_pvt *p = ast->tech_pvt;
ast_mutex_lock(&p->lock);
@@ -2866,24 +2902,12 @@
#ifdef OSP_SUPPORT
time(&p->ospstart);
#endif
-
- codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
- if (codec) {
- fmt=ast_getformatbyname(codec);
- if (fmt) {
- ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
- if (p->jointcapability & fmt) {
- p->jointcapability &= fmt;
- p->capability &= fmt;
- } else
- ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
- } else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
- }
+ try_suggested_sip_codec(p);
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
- res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 1);
+ res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_RELIABLE);
}
ast_mutex_unlock(&p->lock);
return res;
@@ -2907,7 +2931,7 @@
if (p->rtp) {
/* If channel is not up, activate early media session */
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(p, SIP_PROGRESS_SENT);
}
time(&p->lastrtptx);
@@ -2922,7 +2946,7 @@
if (p->vrtp) {
/* Activate video early media */
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(p, SIP_PROGRESS_SENT);
}
time(&p->lastrtptx);
@@ -3065,7 +3089,7 @@
break;
case AST_CONTROL_PROGRESS:
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(p, SIP_PROGRESS_SENT);
break;
}
@@ -4553,7 +4577,7 @@
}
/*! \brief Base transmit response function */
-static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req, int reliable)
+static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
int seqno = 0;
@@ -4579,7 +4603,7 @@
/*! \brief Transmit response, no retransmits */
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req)
{
- return __transmit_response(p, msg, req, 0);
+ return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
}
/*! \brief Transmit response, no retransmits */
@@ -4589,13 +4613,15 @@
respprep(&resp, p, msg, req);
append_date(&resp);
add_header(&resp, "Unsupported", unsupported);
- return send_response(p, &resp, 0, 0);
-}
-
-/*! \brief Transmit response, Make sure you get a reply */
-static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal)
-{
- return __transmit_response(p, msg, req, fatal ? 2 : 1);
+ return send_response(p, &resp, XMIT_UNRELIABLE, 0);
+}
+
+/*! \brief Transmit response, Make sure you get an ACK
+ This is only used for responses to INVITEs, where we need to make sure we get an ACK
+*/
+static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req)
+{
+ return __transmit_response(p, msg, req, XMIT_CRITICAL);
}
/*! \brief Append date to SIP message */
@@ -4619,11 +4645,11 @@
append_date(&resp);
add_header_contentLength(&resp, 0);
add_blank_header(&resp);
- return send_response(p, &resp, 0, 0);
+ return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
/*! \brief Append Accept header, content length before transmitting response */
-static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, int reliable)
+static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
respprep(&resp, p, msg, req);
@@ -4634,7 +4660,7 @@
}
/*! \brief Respond with authorization request */
-static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *randdata, int reliable, const char *header, int stale)
+static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
{
struct sip_request resp;
char tmp[256];
@@ -4945,7 +4971,7 @@
}
/*! \brief Used for 200 OK and 183 early media */
-static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
+static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
int seqno;
@@ -4955,11 +4981,12 @@
}
respprep(&resp, p, msg, req);
if (p->rtp) {
+ try_suggested_sip_codec(p);
add_sdp(&resp, p);
} else {
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
}
- return send_response(p, &resp, retrans, seqno);
+ return send_response(p, &resp, reliable, seqno);
}
/*! \brief Parse first line of incoming SIP request */
@@ -6042,7 +6069,7 @@
}
/*! \brief Transmit generic SIP request */
-static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch)
+static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
reqprep(&resp, p, sipmethod, seqno, newbranch);
@@ -6052,7 +6079,7 @@
}
/*! \brief Transmit SIP request, auth added */
-static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch)
+static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
@@ -6571,7 +6598,7 @@
*/
static int check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
const char *secret, const char *md5secret, int sipmethod,
- char *uri, int reliable, int ignore)
+ char *uri, enum xmittype reliable, int ignore)
{
const char *response = "407 Proxy Authentication Required";
const char *reqheader = "Proxy-Authorization";
@@ -6855,7 +6882,7 @@
} else {
ast_copy_flags(p, peer, SIP_NAT);
transmit_response(p, "100 Trying", req);
- if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, 0, ignore))) {
+ if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE, ignore))) {
sip_cancel_destroy(p);
switch (parse_register_contact(p, peer, req)) {
case PARSE_REGISTER_FAILED:
@@ -7480,7 +7507,7 @@
\return 0 on success, -1 on failure, and 1 on challenge sent
-2 on authentication error from chedck_auth()
*/
-static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen)
+static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen)
{
struct sip_user *user = NULL;
struct sip_peer *peer;
@@ -7773,7 +7800,7 @@
/*! \brief Find user
If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced
*/
-static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore)
+static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, int ignore)
{
return check_user_full(p, req, sipmethod, uri, reliable, sin, ignore, NULL, 0);
}
@@ -9956,6 +9983,8 @@
{
char tmp[256];
char *s, *e;
+ char *domain;
+
ast_copy_string(tmp, get_header(req, "Contact"), sizeof(tmp));
s = get_in_brackets(tmp);
e = strchr(s, ';');
@@ -9972,16 +10001,25 @@
ast_string_field_build(p->owner, call_forward, "SIP/%s", s);
} else {
e = strchr(tmp, '@');
- if (e)
+ if (e) {
*e = '\0';
+ e++;
+ domain = e;
+ } else {
+ /* No username part */
+ domain = tmp;
+ }
e = strchr(tmp, '/');
if (e)
*e = '\0';
if (!strncasecmp(s, "sip:", 4))
s += 4;
- ast_log(LOG_DEBUG, "Found 302 Redirect to extension '%s'\n", s);
- if (p->owner)
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "Received 302 Redirect to extension '%s' (domain %s)\n", s, domain);
+ if (p->owner) {
+ pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain);
ast_string_field_set(p->owner, call_forward, s);
+ }
}
}
@@ -9990,7 +10028,7 @@
{
/* Go ahead and send bye at this point */
if (ast_test_flag(p, SIP_PENDINGBYE)) {
- transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
ast_set_flag(p, SIP_NEEDDESTROY);
ast_clear_flag(p, SIP_NEEDREINVITE);
} else if (ast_test_flag(p, SIP_NEEDREINVITE)) {
@@ -10040,12 +10078,13 @@
break;
case 183: /* Session progress */
sip_cancel_destroy(p);
+ /* Ignore 183 Session progress without SDP */
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
process_sdp(p, req);
- }
- if (!ignore && p->owner) {
- /* Queue a progress frame */
- ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+ if (!ignore && p->owner) {
+ /* Queue a progress frame */
+ ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+ }
}
break;
case 200: /* 200 OK on invite - someone's answering our call */
@@ -10082,13 +10121,13 @@
ast_set_flag(p, SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
- transmit_request(p, SIP_ACK, seqno, 0, 1);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 1);
check_pendings(p);
break;
case 407: /* Proxy authentication */
case 401: /* Www auth */
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
if (p->options)
p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
@@ -10108,7 +10147,7 @@
break;
case 403: /* Forbidden */
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for INVITE to '%s'\n", get_header(&p->initreq, "From"));
if (!ignore && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
@@ -10116,7 +10155,7 @@
ast_set_flag(p, SIP_ALREADYGONE);
break;
case 404: /* Not found */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
if (p->owner && !ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(p, SIP_ALREADYGONE);
@@ -10124,7 +10163,7 @@
case 481: /* Call leg does not exist */
/* Could be REFER or INVITE */
ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
break;
case 491: /* Pending */
/* we have to wait a while, then retransmit */
@@ -10355,7 +10394,7 @@
if (peer->pokeexpire > -1)
ast_sched_del(sched, peer->pokeexpire);
if (sipmethod == SIP_INVITE) /* Does this really happen? */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
ast_set_flag(p, SIP_NEEDDESTROY);
/* Try again eventually */
@@ -10546,6 +10585,7 @@
break;
default:
if ((resp >= 300) && (resp < 700)) {
+ /* Fatal response */
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(iabuf, sizeof(iabuf), p->sa.sin_addr));
ast_set_flag(p, SIP_ALREADYGONE);
@@ -10571,7 +10611,7 @@
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_BUSY);
break;
- case 487:
+ case 487: /* Response on INVITE that has been CANCELled */
/* channel now destroyed - dec the inUse counter */
update_call_counter(p, DEC_CALL_LIMIT);
break;
@@ -10602,7 +10642,7 @@
}
/* ACK on invite */
if (sipmethod == SIP_INVITE)
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
ast_set_flag(p, SIP_ALREADYGONE);
if (!p->owner)
ast_set_flag(p, SIP_NEEDDESTROY);
@@ -11173,12 +11213,12 @@
if (!p->lastinvite && !ignore && !p->owner) { /* This is a new INVITE */
/* Handle authentication if this is our first invite */
- res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore);
+ res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin, ignore);
if (res > 0) /* We have challenged the user for auth */
return 0;
if (res < 0) { /* Something failed in authentication */
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
- transmit_response_reliable(p, "403 Forbidden", req, 1);
+ transmit_response_reliable(p, "403 Forbidden", req);
ast_set_flag(p, SIP_NEEDDESTROY);
ast_string_field_free(p, theirtag);
return 0;
@@ -11187,7 +11227,7 @@
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp") ) {
if (process_sdp(p, req)) {
/* Unacceptable codecs */
- transmit_response_reliable(p, "488 Not acceptable here", req, 1);
+ transmit_response_reliable(p, "488 Not acceptable here", req);
ast_set_flag(p, SIP_NEEDDESTROY);
if (option_debug > 1)
@@ -11218,7 +11258,7 @@
if (res) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
- transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 1);
+ transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
ast_set_flag(p, SIP_NEEDDESTROY);
}
return 0;
@@ -11230,10 +11270,10 @@
if (!replace_id && gotdest) { /* No matching extension found */
if (gotdest < 0) {
- transmit_response_reliable(p, "404 Not Found", req, 1);
+ transmit_response_reliable(p, "404 Not Found", req);
update_call_counter(p, DEC_CALL_LIMIT);
} else {
- transmit_response_reliable(p, "484 Address Incomplete", req, 1);
+ transmit_response_reliable(p, "484 Address Incomplete", req);
update_call_counter(p, DEC_CALL_LIMIT);
}
ast_set_flag(p, SIP_NEEDDESTROY);
@@ -11289,18 +11329,39 @@
transmit_response(p, "100 Trying", req);
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
- if (ast_pbx_start(c)) {
+ enum ast_pbx_result res;
+
+ res = ast_pbx_start(c);
+
+ switch(res) {
+ case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
/* Unlock locks so ast_hangup can do its magic */
- ast_channel_unlock(c);
- ast_mutex_unlock(&p->lock);
- c->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
- ast_hangup(c);
- ast_mutex_lock(&p->lock);
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
- transmit_response_reliable(p, "503 Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Unavailable", req);
+ break;
+ case AST_PBX_CALL_LIMIT:
+ ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+ if (ignore)
+ transmit_response(p, "480 Temporarily Unavailable", req);
+ else
+ transmit_response_reliable(p, "480 Temporarily Unavailable", req);
+ break;
+ case AST_PBX_SUCCESS:
+ /* nothing to do */
+ break;
+ }
+
+ if (res) {
+ ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+
+ /* Unlock locks so ast_hangup can do its magic */
+ ast_mutex_unlock(&c->lock);
+ ast_mutex_unlock(&p->lock);
+ ast_hangup(c);
+ ast_mutex_lock(&p->lock);
c = NULL;
}
} else { /* Pickup call in call group */
@@ -11310,7 +11371,7 @@
if (ignore)
transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
else
- transmit_response_reliable(p, "503 Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Unavailable", req);
ast_set_flag(p, SIP_ALREADYGONE);
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&p->lock);
@@ -11351,14 +11412,14 @@
if (ignore)
transmit_response(p, "488 Not Acceptable Here (codec error)", req);
else
- transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req, 1);
+ transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req);
ast_set_flag(p, SIP_NEEDDESTROY);
} else {
ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
- transmit_response_reliable(p, "503 Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Unavailable", req);
ast_set_flag(p, SIP_NEEDDESTROY);
}
}
@@ -12001,7 +12062,7 @@
ast_set_flag(p, SIP_NEEDDESTROY);
if (p->initreq.len > 0) {
if (!ignore)
- transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1);
+ transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
transmit_response(p, "200 OK", req);
return 1;
} else {
@@ -12019,7 +12080,7 @@
char iabuf[INET_ADDRSTRLEN];
if (p->pendinginvite && !ast_test_flag(p, SIP_OUTGOING) && !ignore)
- transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1);
+ transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
copy_request(&p->initreq, req);
check_via(p, req);
@@ -12120,7 +12181,7 @@
mailboxsize = sizeof(mailboxbuf);
}
/* Handle authentication if this is our first subscribe */
- res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, ignore, mailbox, mailboxsize);
+ res = check_user_full(p, req, SIP_SUBSCRIBE, e, XMIT_UNRELIABLE, sin, ignore, mailbox, mailboxsize);
if (res) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to authenticate user %s for SUBSCRIBE\n", get_header(req, "From"));
@@ -12411,7 +12472,7 @@
if (!p->initreq.headers && ast_test_flag(req, SIP_PKT_WITH_TOTAG)) {
/* If this is a first request and it got a to-tag, it is not for us */
if (!ignore && req->method == SIP_INVITE) {
- transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req, 1);
+ transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
/* Will cease to exist after ACK */
} else {
transmit_response(p, "481 Call/Transaction Does Not Exist", req);
@@ -14308,7 +14369,7 @@
}
ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
[... 554 lines stripped ...]
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