[svn-commits] mmichelson: trunk r89120 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Nov 8 15:01:02 CST 2007
Author: mmichelson
Date: Thu Nov 8 15:01:02 2007
New Revision: 89120
URL: http://svn.digium.com/view/asterisk?view=rev&rev=89120
Log:
Merged revisions 89119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines
Rework of the commit I made yesterday to use the already built-in
ast_uri_decode function as opposed to my home-rolled one. Also added
comments.
Thanks to oej for pointing me in the right direction
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=89120&r1=89119&r2=89120
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Nov 8 15:01:02 2007
@@ -4503,17 +4503,6 @@
return res;
}
-static char *translate_escaped_pound(char *exten)
-{
- char *rest, *marker;
- while((marker = strstr(exten, "%23"))) {
- rest = marker + 3;
- *marker++ = '#';
- memmove(marker, rest, strlen(rest) + 1);
- }
- return exten;
-}
-
/*! \brief Initiate a call in the SIP channel
called from sip_request_call (calls from the pbx ) for outbound channels
@@ -4531,6 +4520,7 @@
int needvideo = 0;
int needtext = 0;
char buf[BUFSIZ];
+ char *decoded_exten;
{
const char *my_name; /* pick a good name */
@@ -4648,7 +4638,13 @@
i->owner = tmp;
ast_module_ref(ast_module_info->self);
ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
- ast_copy_string(tmp->exten, translate_escaped_pound(ast_strdupa(i->exten)), sizeof(tmp->exten));
+ /*Since it is valid to have extensions in the dialplan that have unescaped characters in them
+ * we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt
+ * structure so that there aren't issues when forming URI's
+ */
+ decoded_exten = ast_strdupa(i->exten);
+ ast_uri_decode(decoded_exten);
+ ast_copy_string(tmp->exten, decoded_exten, sizeof(tmp->exten));
/* Don't use ast_set_callerid() here because it will
* generate an unnecessary NewCallerID event */
@@ -9600,26 +9596,17 @@
} else {
/* Check the dialplan for the username part of the request URI,
the domain will be stored in the SIPDOMAIN variable
+ Since extensions.conf can have unescaped characters, try matching a decoded
+ uri in addition to the non-decoded uri
Return 0 if we have a matching extension */
- if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) ||
+ char *decoded_uri = ast_strdupa(uri);
+ ast_uri_decode(decoded_uri);
+ if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) || ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) ||
!strcmp(uri, ast_pickup_ext())) {
if (!oreq)
ast_string_field_set(p, exten, uri);
return 0;
- } else { /*Could be trying to match a literal '#'. Try replacing and see if that works.*/
- char *tmpuri = ast_strdupa(uri);
- char *rest, *marker;
- while((marker = strstr(tmpuri, "%23"))) {
- rest = marker + 3;
- *marker++ = '#';
- memmove(marker, rest, strlen(rest) + 1);
- }
- if(ast_exists_extension(NULL, p->context, tmpuri, 1, from) || !strcmp(uri, ast_pickup_ext())) {
- if(!oreq)
- ast_string_field_set(p, exten, uri);
- return 0;
- }
- }
+ }
}
/* Return 1 for pickup extension or overlap dialling support (if we support it) */
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