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authortwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2011-02-07 22:15:27 +0000
committertwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2011-02-07 22:15:27 +0000
commitaf25cea2b0d6f4e6661a29e6e1c59c1779b27649 (patch)
treec5497bc64a1262d81df71ef7d44a43f2044fc189
parent177925ab968f095566a6010292262e938ab44d72 (diff)
Merged revisions 306618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@306619 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c10
1 files changed, 9 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9f95510db..64188592f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -14727,7 +14727,15 @@ static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoi
}
ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
}
-
+
+ if (!strcmp(referdata->replaces_callid, transferer->callid) &&
+ (!sip_cfg.pedanticsipchecking ||
+ (!strcmp(referdata->replaces_callid_fromtag, transferer->tag) &&
+ !strcmp(referdata->replaces_callid_totag, transferer->theirtag)))) {
+ ast_log(LOG_WARNING, "Got an attempt to replace own Call-ID on %s\n", transferer->callid);
+ return -4;
+ }
+
if (!sip_cfg.pedanticsipchecking) {
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
} else {