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make chan_sip able to deal with PBX-level call limit being reached (issue #5131)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2005-10-14 01:14:29 +00:00
parent 2fe49e7a5f
commit 4d1506e64d
1 changed files with 25 additions and 5 deletions
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30
channels/chan_sip.c
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@@ -10344,17 +10344,37 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, "100 Trying", req);
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) {
if (ast_pbx_start(c)) {
enum ast_pbx_result res;
res = ast_pbx_start(c);
switch (res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
transmit_response_reliable(p, "503 Unavailable", req, 1);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
if (ignore)
transmit_response(p, "480 Temporarily Unavailable", req);
else
transmit_response_reliable(p, "480 Temporarily Unavailable", req, 1);
break;
case AST_PBX_SUCCESS:
/* nothing to do */
break;
}
if (res) {
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&c->lock);
ast_mutex_unlock(&p->lock);
ast_hangup(c);
ast_mutex_lock(&p->lock);
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
transmit_response_reliable(p, "503 Unavailable", req, 1);
c = NULL;
}
} else {