[asterisk-commits] mjordan: branch 13 r424622 - in /branches/13: ./ channels/pjsip/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Oct 5 19:52:22 CDT 2014
Author: mjordan
Date: Sun Oct 5 19:52:19 2014
New Revision: 424622
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=424622
Log:
pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels
Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.
This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.
#SIPit31
ASTERISK-24382 #close
Reported by: Matt Jordan
........
Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12
Modified:
branches/13/ (props changed)
branches/13/channels/pjsip/dialplan_functions.c
Propchange: branches/13/
------------------------------------------------------------------------------
Binary property 'branch-12-merged' - no diff available.
Modified: branches/13/channels/pjsip/dialplan_functions.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/channels/pjsip/dialplan_functions.c?view=diff&rev=424622&r1=424621&r2=424622
==============================================================================
--- branches/13/channels/pjsip/dialplan_functions.c (original)
+++ branches/13/channels/pjsip/dialplan_functions.c Sun Oct 5 19:52:19 2014
@@ -707,7 +707,7 @@
/* Sanity check */
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
- ast_log(LOG_ERROR, "Cannot call %s on a non-PJSIP channel\n", cmd);
+ ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return 0;
}
@@ -866,6 +866,11 @@
return -1;
}
+ if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+ ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
+ return -1;
+ }
+
channel = ast_channel_tech_pvt(chan);
if (!strcmp(data, "audio")) {
@@ -889,6 +894,11 @@
return -1;
}
+ if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+ ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
+ return -1;
+ }
+
channel = ast_channel_tech_pvt(chan);
mdata.session = channel->session;
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