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Merged revisions 158053 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines

Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.

(closes issue #13867)
Reported by: still_nsk
Patches:
      13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@158066 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
mmichelson 2008-11-20 17:39:06 +00:00
parent d93047742f
commit 056bae58f4
2 changed files with 7 additions and 1 deletions
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3
apps/app_dial.c
View File

@ -1818,6 +1818,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
/* Again, keep going even if there's an error */
ast_debug(1, "ast call on peer returned %d\n", res);
ast_verb(3, "Couldn't call %s\n", numsubst);
if (tc->hangupcause) {
chan->hangupcause = tc->hangupcause;
}
ast_hangup(tc);
tc = NULL;
ast_free(tmp);

5
channels/chan_sip.c
View File

@ -4870,8 +4870,11 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
res = update_call_counter(p, INC_CALL_RINGING);
if (res == -1)
if (res == -1) {
return res;
} else {
ast->hangupcause = AST_CAUSE_USER_BUSY;
}
p->callingpres = ast->cid.cid_pres;
p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);