]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Fix deadlock during dialplan reload.
authorRichard Mudgett <rmudgett@digium.com>
Wed, 9 Nov 2011 20:37:53 +0000 (20:37 +0000)
committerRichard Mudgett <rmudgett@digium.com>
Wed, 9 Nov 2011 20:37:53 +0000 (20:37 +0000)
Another deadlock between the conlock/hints and channels/channel locking
orders.

* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().

(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
      sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
      ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344268 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 92db90401a143ae250bb9032ff80afa5988d4ad6..6ff92a7acb66418d3bccc84bacaf7dbc65f3cd9a 100644 (file)
@@ -6937,7 +6937,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
        format_t needvideo = 0;
        int needtext = 0;
        char buf[SIPBUFSIZE];
-       char *decoded_exten;
+       char *exten;
 
        {
                const char *my_name;    /* pick a good name */
@@ -7077,14 +7077,15 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
         * 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
         */
-       if (ast_exists_extension(NULL, i->context, i->exten, 1, i->cid_num)) {
-               /* encoded in dialplan, so keep extension encoded */
-               ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
-       } else {
-               decoded_exten = ast_strdupa(i->exten);
-               ast_uri_decode(decoded_exten);
-               ast_copy_string(tmp->exten, decoded_exten, sizeof(tmp->exten));
+       exten = ast_strdupa(i->exten);
+       sip_pvt_unlock(i);
+       ast_channel_unlock(tmp);
+       if (!ast_exists_extension(NULL, i->context, i->exten, 1, i->cid_num)) {
+               ast_uri_decode(exten);
        }
+       ast_channel_lock(tmp);
+       sip_pvt_lock(i);
+       ast_copy_string(tmp->exten, exten, sizeof(tmp->exten));
 
        /* Don't use ast_set_callerid() here because it will
         * generate an unnecessary NewCallerID event  */