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authorDavid Vossel <dvossel@digium.com>2009-04-21 20:28:37 +0000
committerDavid Vossel <dvossel@digium.com>2009-04-21 20:28:37 +0000
commit00d7c4fefc78a5af0af4e41516cd9911dfd232c7 (patch)
tree71452e9fc681a823dadd87ffae42a98e9bb02f81 /channels
parent6bb2b6c096cc8c56d0a01fb17d9723b8c5b92d0e (diff)
Fixes segfault when switching UDP to TCP in sip.conf after reload.
If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found. (closes issue #14727) Reported by: pj Tested by: dvossel Review: http://reviewboard.digium.com/r/229/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c23
1 files changed, 15 insertions, 8 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c556d91ce..29b122f8b 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3313,14 +3313,18 @@ static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len)
if (p->socket.tcptls_session)
ast_mutex_lock(&p->socket.tcptls_session->lock);
- if (p->socket.type & SIP_TRANSPORT_UDP)
+ if (p->socket.type & SIP_TRANSPORT_UDP) {
res = sendto(p->socket.fd, data->str, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
- else {
- if (p->socket.tcptls_session->f)
+ } else if (p->socket.tcptls_session) {
+ if (p->socket.tcptls_session->f) {
res = ast_tcptls_server_write(p->socket.tcptls_session, data->str, len);
- else
+ } else {
ast_debug(2, "No p->socket.tcptls_session->f len=%d\n", len);
- }
+ }
+ } else {
+ ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
+ return XMIT_ERROR;
+ }
if (p->socket.tcptls_session)
ast_mutex_unlock(&p->socket.tcptls_session->lock);
@@ -10499,7 +10503,7 @@ static void state_notify_build_xml(int state, int full, const char *exten, const
ast_str_append(tmp, 0, "<dialog id=\"%s\">", exten);
}
ast_str_append(tmp, 0, "<state>%s</state>\n", statestring);
- if (state == AST_EXTENSION_ONHOLD) { //todohere, this seems weird
+ if (state == AST_EXTENSION_ONHOLD) {
ast_str_append(tmp, 0, "<local>\n<target uri=\"%s\">\n"
"<param pname=\"+sip.rendering\" pvalue=\"no\"/>\n"
"</target>\n</local>\n", mto);
@@ -23276,8 +23280,10 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
else
ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
- if (!peer->socket.type) /*!< The first transport listed should be used for outgoing */
+ if (!peer->socket.type) { /*!< The first transport listed should be used for outgoing */
peer->socket.type = peer->transports;
+ peer->socket.fd = -1;
+ }
}
} else if (realtime && !strcasecmp(v->name, "regseconds")) {
ast_get_time_t(v->value, &regseconds, 0, NULL);
@@ -23577,9 +23583,10 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
if (!peer->socket.type) {
/* Set default set of transports */
- peer->transports = default_transports;
+ peer->transports = default_transports;
/* Set default primary transport */
peer->socket.type = default_primary_transport;
+ peer->socket.fd = -1;
}
if (fullcontact->used > 0) {