diff options
| author | Jason Parker <jparker@digium.com> | 2011-01-31 23:08:38 +0000 |
|---|---|---|
| committer | Jason Parker <jparker@digium.com> | 2011-01-31 23:08:38 +0000 |
| commit | 6908539952b758b9e555c2f76beffdc36675734a (patch) | |
| tree | 54ceac326ec2776a05e9b306a90c40a2a6e5c300 | |
| parent | 14c158564586f99daf27cbb734af167d1fba6eff (diff) | |
Merged revisions 305254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines
Merged revisions 305253 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines
Merged revisions 305252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines
Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
chan_iax2 and other channel drivers already had code to prevent this. The
attempt that app_dial was making to prevent it was not correct, so I fixed that.
(closes issue #18371)
Reported by: gbour
Patches:
18371.patch uploaded by gbour (license 1162)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | apps/app_dial.c | 2 | ||||
| -rw-r--r-- | channels/chan_sip.c | 6 |
2 files changed, 7 insertions, 1 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index a1522a81a..581a88170 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -1944,7 +1944,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast struct ast_dialed_interface *di; AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces; num_dialed++; - if (!number) { + if (ast_strlen_zero(number)) { ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n"); goto out; } diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d0c0fffa3..a33a0ddd1 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -25347,6 +25347,12 @@ static struct ast_channel *sip_request_call(const char *type, format_t format, c } ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat)); + if (ast_strlen_zero(dest)) { + ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n"); + *cause = AST_CAUSE_CHANNEL_UNACCEPTABLE; + return NULL; + } + if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE, NULL))) { ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest); *cause = AST_CAUSE_SWITCH_CONGESTION; |