[asterisk-commits] jrose: trunk r319939 - in /trunk: ./ channels/ channels/sip/include/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri May 20 08:42:21 CDT 2011
Author: jrose
Date: Fri May 20 08:42:15 2011
New Revision: 319939
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=319939
Log:
Merged revisions 319938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
(closes issue #18344)
Reported by: danimal
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1223/
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/channels/sip/include/sip.h
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=319939&r1=319938&r2=319939
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri May 20 08:42:15 2011
@@ -13524,6 +13524,18 @@
return TRUE;
}
+/*! \brief parse uri in a way that allows semicolon stripping if legacy mode is enabled */
+static int parse_uri_legacy_check(char *uri, const char *scheme, char **user, char **pass, char **domain, char **transport) {
+ int ret = parse_uri(uri, scheme, user, pass, domain, transport);
+ if (sip_cfg.legacy_useroption_parsing) { /* if legacy mode is active, strip semis from the user field */
+ char *p;
+ if ((p = strchr(uri, (int)';'))) {
+ *p = '\0';
+ }
+ }
+ return ret;
+}
+
static int __set_address_from_contact(const char *fullcontact, struct ast_sockaddr *addr, int tcp)
{
char *domain, *transport;
@@ -13541,7 +13553,7 @@
* We still need to be able to send to the remote agent through the proxy.
*/
- if (parse_uri(contact, "sip:,sips:", &contact, NULL, &domain,
+ if (parse_uri_legacy_check(contact, "sip:,sips:", &contact, NULL, &domain,
&transport)) {
ast_log(LOG_WARNING, "Invalid contact uri %s (missing sip: or sips:), attempting to use anyway\n", fullcontact);
}
@@ -13670,7 +13682,7 @@
ast_string_field_build(pvt, our_contact, "<%s>", curi);
/* Make sure it's a SIP URL */
- if (parse_uri(curi, "sip:,sips:", &curi, NULL, &domain, &transport)) {
+ if (parse_uri_legacy_check(curi, "sip:,sips:", &curi, NULL, &domain, &transport)) {
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:/sips:) trying to use anyway\n");
}
@@ -14371,7 +14383,7 @@
c = get_in_brackets(tmp);
c = remove_uri_parameters(c);
- if (parse_uri(c, "sip:,sips:", &name, &dummy, &domain, NULL)) {
+ if (parse_uri_legacy_check(c, "sip:,sips:", &name, &dummy, &domain, NULL)) {
ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_sockaddr_stringify_addr(addr));
return -1;
}
@@ -14952,7 +14964,7 @@
uri = ast_strdupa(get_in_brackets(tmp));
- if (parse_uri(uri, "sip:,sips:", &uri, &dummy, &domain, NULL)) {
+ if (parse_uri_legacy_check(uri, "sip:,sips:", &uri, &dummy, &domain, NULL)) {
ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", uri);
return SIP_GET_DEST_INVALID_URI;
}
@@ -14977,7 +14989,7 @@
ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf));
if (!ast_strlen_zero(tmpf)) {
from = get_in_brackets(tmpf);
- if (parse_uri(from, "sip:,sips:", &from, NULL, &domain, NULL)) {
+ if (parse_uri_legacy_check(from, "sip:,sips:", &from, NULL, &domain, NULL)) {
ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", from);
return SIP_GET_DEST_INVALID_URI;
}
@@ -15388,7 +15400,7 @@
ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp));
c = get_in_brackets(tmp);
- if (parse_uri(c, "sip:,sips:", &c, NULL, &a, NULL)) {
+ if (parse_uri_legacy_check(c, "sip:,sips:", &c, NULL, &a, NULL)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header in Also: transfer (%s)?\n", c);
return -1;
}
@@ -15764,7 +15776,7 @@
ast_string_field_set(p, from, of);
/* ignore all fields but name */
- if (parse_uri(of, "sip:,sips:", &of, &dummy, &domain, NULL)) {
+ if (parse_uri_legacy_check(of, "sip:,sips:", &of, &dummy, &domain, NULL)) {
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
@@ -17619,6 +17631,7 @@
ast_cli(a->fd, " Regexten on Qualify: %s\n", AST_CLI_YESNO(sip_cfg.regextenonqualify));
ast_cli(a->fd, " Trust RPID: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_TRUSTRPID)));
ast_cli(a->fd, " Send RPID: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_SENDRPID)));
+ ast_cli(a->fd, " Legacy userfield parse: %s\n", AST_CLI_YESNO(sip_cfg.legacy_useroption_parsing));
ast_cli(a->fd, " Caller ID: %s\n", default_callerid);
if ((default_fromdomainport) && (default_fromdomainport != STANDARD_SIP_PORT)) {
ast_cli(a->fd, " From: Domain: %s:%d\n", default_fromdomain, default_fromdomainport);
@@ -27303,6 +27316,7 @@
sip_cfg.regcontext[0] = '\0';
sip_set_default_format_capabilities(sip_cfg.caps);
sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
+ sip_cfg.legacy_useroption_parsing = DEFAULT_LEGACY_USEROPTION_PARSING;
sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING;
sip_cfg.notifycid = DEFAULT_NOTIFYCID;
sip_cfg.notifyhold = FALSE; /*!< Keep track of hold status for a peer */
@@ -27576,6 +27590,8 @@
ast_copy_string(sip_cfg.regcontext, v->value, sizeof(sip_cfg.regcontext));
} else if (!strcasecmp(v->name, "regextenonqualify")) {
sip_cfg.regextenonqualify = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "legacy_useroption_parsing")) {
+ sip_cfg.legacy_useroption_parsing = ast_true(v->value);
} else if (!strcasecmp(v->name, "callerid")) {
ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
} else if (!strcasecmp(v->name, "mwi_from")) {
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=319939&r1=319938&r2=319939
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Fri May 20 08:42:15 2011
@@ -212,6 +212,7 @@
#define DEFAULT_ALWAYSAUTHREJECT TRUE /*!< Don't reject authentication requests always */
#define DEFAULT_AUTH_OPTIONS FALSE
#define DEFAULT_REGEXTENONQUALIFY FALSE
+#define DEFAULT_LEGACY_USEROPTION_PARSING FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
#ifndef DEFAULT_USERAGENT
@@ -683,6 +684,7 @@
int allow_external_domains; /*!< Accept calls to external SIP domains? */
int callevents; /*!< Whether we send manager events or not */
int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
+ int legacy_useroption_parsing; /*!< Whether to strip useroptions in URI via semicolons */
int matchexternaddrlocally; /*!< Match externaddr/externhost setting against localnet setting */
char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
unsigned int disallowed_methods; /*!< methods that we should never try to use */
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=319939&r1=319938&r2=319939
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Fri May 20 08:42:15 2011
@@ -432,6 +432,14 @@
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
+;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
+ ; in the user field of a sip URI, the field be truncated
+ ; at the first semicolon seen. This effectively makes
+ ; semicolon a non-usable character for peer names, extensions,
+ ; and maybe other, less tested things. This can be useful
+ ; for improving compatability with devices that like to use
+ ; user options for whatever reason. The behavior is similar to
+ ; how SIP URI's were typically handled in 1.6.2, hence the name.
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
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