[svn-commits] oej: branch oej/codename-pineapple r44728 -
/team/oej/codename-pineapple/chan...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 7 12:37:38 MST 2006
Author: oej
Date: Sat Oct 7 14:37:37 2006
New Revision: 44728
URL: http://svn.digium.com/view/asterisk?rev=44728&view=rev
Log:
This patch just feels great! What a feeling on a Saturday evening.
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
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=44728&r1=44727&r2=44728&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Sat Oct 7 14:37:37 2006
@@ -479,7 +479,6 @@
#define DEFAULT_ALLOW_EXT_DOM TRUE
#define DEFAULT_REALM "asterisk"
#define DEFAULT_NOTIFYRINGING TRUE
-#define DEFAULT_PEDANTIC FALSE
#define DEFAULT_AUTOCREATEPEER FALSE
#define DEFAULT_QUALIFY FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
@@ -512,7 +511,6 @@
static int global_notifyringing; /*!< Send notifications on ringing */
static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
-static int pedanticsipchecking; /*!< Extra checking ? Default off */
static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
static int global_relaxdtmf; /*!< Relax DTMF */
static int global_rtptimeout; /*!< Time out call if no RTP */
@@ -3831,16 +3829,13 @@
* one afterwards. If you shouldn't do it, what absolute idiot decided it was
* a good idea to say you can do it, and if you can do it, why in the hell would.
* you say you shouldn't.
- * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
- * and we always allow spaces after that for compatibility.
*/
for (pass = 0; name && pass < 2;pass++) {
int x, len = strlen(name);
for (x=*start; x<req->headers; x++) {
if (!strncasecmp(req->header[x], name, len)) {
char *r = req->header[x] + len; /* skip name */
- if (pedanticsipchecking)
- r = ast_skip_blanks(r);
+ r = ast_skip_blanks(r);
if (*r == ':') {
*start = x+1;
@@ -4143,38 +4138,34 @@
if (!callid || !to || !from || !cseq) /* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
return NULL; /* Invalid packet */
- if (pedanticsipchecking) {
- /* 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.
- For Asterisk to behave correctly, you need to turn on pedanticsipchecking
- in sip.conf
- */
- 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));
-
- 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);
- }
+ /* 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)))
+ ast_set_flag(req, SIP_PKT_WITH_TOTAG); /* Used in handle_request/response */
+ 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_mutex_lock(&iflock);
for (p = iflist; p; p = p->next) {
- /* In pedantic, we do not want packets with bad syntax to be connected to a PVT */
+ /* 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));
else
found = (!strcmp(p->callid, callid) &&
- (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ;
+ (!tag || ast_strlen_zero(p->theirtag) || !strcmp(p->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);
/* If we get a new request within an existing to-tag - check the to tag as well */
- if (pedanticsipchecking && found && req->method != SIP_RESPONSE) { /* SIP Request */
+ if (found && req->method != SIP_RESPONSE) { /* SIP Request */
if (p->tag[0] == '\0' && totag[0]) {
/* We have no to tag, but they have. Wrong dialog */
found = FALSE;
@@ -4278,8 +4269,7 @@
return 0;
}
-/*! \brief Parse multiline SIP headers into one header
- This is enabled if pedanticsipchecking is enabled */
+/*! \brief Parse multiline SIP headers into one header */
static int lws2sws(char *msgbuf, int len)
{
int h = 0, t = 0;
@@ -6379,12 +6369,10 @@
else /* Save for any further attempts */
ast_string_field_set(p, fromname, n);
- if (pedanticsipchecking) {
- ast_uri_encode(n, tmp, sizeof(tmp), 0);
- n = tmp;
- ast_uri_encode(l, tmp2, sizeof(tmp2), 0);
- l = tmp2;
- }
+ ast_uri_encode(n, tmp, sizeof(tmp), 0);
+ n = tmp;
+ ast_uri_encode(l, tmp2, sizeof(tmp2), 0);
+ l = tmp2;
if ((ourport != 5060) && ast_strlen_zero(p->fromdomain)) /* Needs to be 5060 */
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip)), ourport, p->tag);
@@ -6400,10 +6388,8 @@
ast_build_string(&invite, &invite_max, "sip:");
if (!ast_strlen_zero(p->username)) {
n = p->username;
- if (pedanticsipchecking) {
- ast_uri_encode(n, tmp, sizeof(tmp), 0);
- n = tmp;
- }
+ ast_uri_encode(n, tmp, sizeof(tmp), 0);
+ n = tmp;
ast_build_string(&invite, &invite_max, "%s@", n);
}
ast_build_string(&invite, &invite_max, "%s", p->tohost);
@@ -7926,8 +7912,7 @@
*t = '\0';
ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp));
- if (pedanticsipchecking)
- ast_uri_decode(tmp);
+ ast_uri_decode(tmp);
c = get_in_brackets(tmp);
c = strsep(&c, ";"); /* Ditch ;user=phone */
@@ -8126,8 +8111,7 @@
if (req->rlPart2)
ast_copy_string(tmp, req->rlPart2, sizeof(tmp));
- if (pedanticsipchecking)
- ast_uri_decode(tmp);
+ ast_uri_decode(tmp);
uri = get_in_brackets(tmp);
@@ -8140,8 +8124,7 @@
/* Now find the From: caller ID and name */
ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf));
if (!ast_strlen_zero(tmpf)) {
- if (pedanticsipchecking)
- ast_uri_decode(tmpf);
+ ast_uri_decode(tmpf);
from = get_in_brackets(tmpf);
} else {
from = NULL;
@@ -8246,7 +8229,7 @@
(With a forking SIP proxy, several call legs share the
call id, but have different tags)
*/
- if (pedanticsipchecking && (strcmp(fromtag, sip_pvt_ptr->theirtag) || strcmp(totag, ourtag)))
+ if (strcmp(fromtag, sip_pvt_ptr->theirtag) || strcmp(totag, ourtag))
match = 0;
if (!match) {
@@ -8303,8 +8286,7 @@
}
h_refer_to = ast_strdupa(p_refer_to);
refer_to = get_in_brackets(h_refer_to);
- if (pedanticsipchecking)
- ast_uri_decode(refer_to);
+ ast_uri_decode(refer_to);
if (strncasecmp(refer_to, "sip:", 4)) {
ast_log(LOG_WARNING, "Can't transfer to non-sip: URI. (Refer-to: %s)?\n", refer_to);
@@ -8316,8 +8298,7 @@
if ((p_referred_by = get_header(req, "Referred-By"))) {
char *lessthan;
h_referred_by = ast_strdupa(p_referred_by);
- if (pedanticsipchecking)
- ast_uri_decode(h_referred_by);
+ ast_uri_decode(h_referred_by);
/* Store referrer's caller ID name */
ast_copy_string(referdata->referred_by_name, h_referred_by, sizeof(referdata->referred_by_name));
@@ -8373,12 +8354,8 @@
ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
}
- if (option_debug > 1) {
- if (!pedanticsipchecking)
- ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
- else
- ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
- }
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
}
}
@@ -8445,8 +8422,7 @@
ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp));
c = get_in_brackets(tmp);
- if (pedanticsipchecking)
- ast_uri_decode(c);
+ ast_uri_decode(c);
if (strncmp(c, "sip:", 4)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header in Also: transfer (%s)?\n", c);
@@ -8640,8 +8616,7 @@
t++;
*t = '\0';
ast_copy_string(from, get_header(req, "From"), sizeof(from)); /* XXX bug in original code, overwrote string */
- if (pedanticsipchecking)
- ast_uri_decode(from);
+ ast_uri_decode(from);
/* XXX here tries to map the username for invite things */
memset(calleridname, 0, sizeof(calleridname));
get_calleridname(from, calleridname, sizeof(calleridname));
@@ -9954,7 +9929,6 @@
ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
ast_cli(fd, " DNS SRV lookup: %s\n", srvlookup ? "Yes" : "No");
- ast_cli(fd, " Pedantic SIP support: %s\n", pedanticsipchecking ? "Yes" : "No");
ast_cli(fd, " Reg. min duration %d secs\n", min_expiry);
ast_cli(fd, " Reg. max duration: %d secs\n", max_expiry);
ast_cli(fd, " Reg. default duration: %d secs\n", default_expiry);
@@ -12671,8 +12645,8 @@
/* This is a call to ourself. Send ourselves an error code and stop
processing immediately, as SIP really has no good mechanism for
being able to call yourself */
- /* If pedantic is on, we need to check the tags. If they're different, this is
- in fact a forked call through a SIP proxy somewhere. */
+ /* we need to check the tags. If they're different, this is
+ in fact a forked call through a SIP proxy somewhere. */
transmit_response(p, "482 Loop Detected", req);
/* We do NOT destroy p here, so that our response will be accepted */
return 0;
@@ -14117,22 +14091,20 @@
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);
- if (pedanticsipchecking) {
- /* If this is a request packet without a from tag, it's not
- correct according to RFC 3261 */
- /* Check if this a new request in a new dialog with a totag already attached to it,
- RFC 3261 - section 12.2 - and we don't want to mess with recovery */
- 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 (!ast_test_flag(req, SIP_PKT_IGNORE) && req->method == SIP_INVITE) {
- transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
- /* Will cease to exist after ACK */
- } else if (req->method != SIP_ACK) {
- transmit_response(p, "481 Call/Transaction Does Not Exist", req);
- sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- }
- return res;
- }
+ /* If this is a request packet without a from tag, it's not
+ correct according to RFC 3261 */
+ /* Check if this a new request in a new dialog with a totag already attached to it,
+ RFC 3261 - section 12.2 - and we don't want to mess with recovery */
+ 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 (!ast_test_flag(req, SIP_PKT_IGNORE) && req->method == SIP_INVITE) {
+ transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
+ /* Will cease to exist after ACK */
+ } else if (req->method != SIP_ACK) {
+ transmit_response(p, "481 Call/Transaction Does Not Exist", req);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ }
+ return res;
}
/* Handle various incoming SIP methods in requests */
@@ -14236,8 +14208,7 @@
req.len = res;
if(sip_debug_test_addr(&sin)) /* Set the debug flag early on packet level */
ast_set_flag(&req, SIP_PKT_DEBUG);
- if (pedanticsipchecking)
- req.len = lws2sws(req.data, req.len); /* Fix multiline headers */
+ req.len = lws2sws(req.data, req.len); /* Fix multiline headers */
if (ast_test_flag(&req, SIP_PKT_DEBUG))
ast_verbose("\n<-- SIP read from %s:%d: \n%s\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), req.data);
@@ -15593,7 +15564,6 @@
compactheaders = DEFAULT_COMPACTHEADERS;
global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
global_regattempts_max = 0;
- pedanticsipchecking = DEFAULT_PEDANTIC;
global_mwitime = DEFAULT_MWITIME;
autocreatepeer = DEFAULT_AUTOCREATEPEER;
global_allowguest = DEFAULT_ALLOWGUEST;
@@ -15740,8 +15710,6 @@
autocreatepeer = ast_true(v->value);
} else if (!strcasecmp(v->name, "srvlookup")) {
srvlookup = ast_true(v->value);
- } else if (!strcasecmp(v->name, "pedantic")) {
- pedanticsipchecking = ast_true(v->value);
} else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
max_expiry = atoi(v->value);
if (max_expiry < 1)
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