[asterisk-commits] oej: branch oej/pgtips-srv-and-outbound-stuff-1.8 r421664 - in /team/oej/pgti...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Aug 21 04:41:30 CDT 2014
Author: oej
Date: Thu Aug 21 04:41:13 2014
New Revision: 421664
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=421664
Log:
Trying to understand the old transmit_register() function. Some of the things here
are a mystery. In order to add proper DNS srv failover and lookup a lot of stuff
needs to be re-ordered and changed.
Also implemented a flag that indicates that someone configured a port number. In
that case, SRV should be disabled. That part is a bug fix I think.
Modified:
team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/chan_sip.c
team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/config_parser.c
team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/include/sip.h
team/oej/pgtips-srv-and-outbound-stuff-1.8/configs/sip.conf.sample
Modified: team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/chan_sip.c?view=diff&rev=421664&r1=421663&r2=421664
==============================================================================
--- team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/chan_sip.c (original)
+++ team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/chan_sip.c Thu Aug 21 04:41:13 2014
@@ -5753,8 +5753,9 @@
peername2 = ast_strdupa(opeer);
AST_NONSTANDARD_RAW_ARGS(hostport, peername2, ':');
- if (hostport.port)
- dialog->portinuri = 1;
+ if (hostport.port) {
+ dialog->portinuri = TRUE;
+ }
dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
@@ -5804,7 +5805,7 @@
/* Section 4.2 of RFC 3263 specifies that if a port number is specified, then
* an A record lookup should be used instead of SRV.
*/
- if (!hostport.port && sip_cfg.srvlookup) {
+ if (!dialog->portinuri && sip_cfg.srvlookup) {
if (dialog->srvcontext) {
ast_srv_context_free_list(dialog->srvcontext);
}
@@ -13840,12 +13841,19 @@
snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(r->transport), get_srv_protocol(r->transport)); /* have to use static get_transport function */
r->us.ss.ss_family = get_address_family_filter(r->transport); /* Filter address family */
+#ifdef DISABLE_DUAL_LOOKUP
+ /* OEJ: Disabling this lookup, since create_addr is better at using DNS SRV records for
+ now. Remind me wy we do the SRV here instead of in the dialog, since we keep the dialog
+ anyway.
+ */
/* No point in doing a DNS lookup of the register hostname if we're just going to
* end up using an outbound proxy. obproxy_get is safe to call with either of r->call
* or peer NULL. Since we're only concerned with its existence, we're not going to
* bother getting a ref to the proxy*/
if (!obproxy_get(r->call, peer)) {
+ /* Why are we doing this when create_addr is doing it for us? */
registry_addref(r, "add reg ref for dnsmgr");
+ /* If we have a configured port number, do not do SRV lookups */
ast_dnsmgr_lookup_cb(peer ? peer->tohost : r->hostname, &r->us, &r->dnsmgr, sip_cfg.srvlookup ? transport : NULL, on_dns_update_registry, r);
if (!r->dnsmgr) {
/*dnsmgr refresh disabled, no reference added! */
@@ -13856,6 +13864,7 @@
peer = unref_peer(peer, "removing peer ref for dnsmgr_lookup");
}
}
+#endif
if (r->call) { /* We have a registration */
if (!auth) {
@@ -13880,6 +13889,9 @@
/* reset tag to consistent value from registry */
ast_string_field_set(p, tag, r->localtag);
+
+ /* If we have a port configured, do not activate SRV record lookup for this host */
+ p->portinuri = r->portconfigured;
if (p->do_history) {
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
@@ -13904,6 +13916,18 @@
}
peer = unref_peer(peer, "unref after find_peer");
}
+ }
+
+ /* Set transport and port so the correct contact is built */
+ set_socket_transport(&p->socket, r->transport);
+ if (r->transport == SIP_TRANSPORT_TLS || r->transport == SIP_TRANSPORT_TCP) {
+ p->socket.port =
+ htons(ast_sockaddr_port(&sip_tcp_desc.local_address));
+ }
+ if (!r->dnsmgr && r->portno) {
+ ast_sockaddr_set_port(&p->sa, r->portno);
+ ast_sockaddr_set_port(&p->recv, r->portno);
+ ast_debug(2, "Confusing code set port to %d\n", r->portno);
}
/* Find address to hostname */
@@ -13929,11 +13953,6 @@
/* Copy back Call-ID in case create_addr changed it */
ast_string_field_set(r, callid, p->callid);
- if (!r->dnsmgr && r->portno) {
- ast_sockaddr_set_port(&p->sa, r->portno);
- ast_sockaddr_set_port(&p->recv, r->portno);
- ast_debug(2, "Confusing code set port to %d\n", r->portno);
- }
if (!ast_strlen_zero(p->fromdomain)) {
portno = (p->fromdomainport) ? p->fromdomainport : STANDARD_SIP_PORT;
} else if (!ast_strlen_zero(r->regdomain)) {
@@ -13968,12 +13987,6 @@
ast_string_field_set(p, exten, r->callback);
}
- /* Set transport and port so the correct contact is built */
- set_socket_transport(&p->socket, r->transport);
- if (r->transport == SIP_TRANSPORT_TLS || r->transport == SIP_TRANSPORT_TCP) {
- p->socket.port =
- htons(ast_sockaddr_port(&sip_tcp_desc.local_address));
- }
/*
check which address we should use in our contact header
@@ -14065,11 +14078,11 @@
initialize_initreq(p, &req);
if (sip_debug_test_pvt(p)) {
- ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
+ ast_verbose("SIP REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT;
r->regattempts++; /* Another attempt */
- ast_debug(4, "REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
+ ast_debug(4, "SIP REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
res = send_request(p, &req, XMIT_CRITICAL, p->ocseq);
dialog_unref(p, "p is finished here at the end of transmit_register");
return res;
Modified: team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/config_parser.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/config_parser.c?view=diff&rev=421664&r1=421663&r2=421664
==============================================================================
--- team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/config_parser.c (original)
+++ team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/config_parser.c Thu Aug 21 04:41:13 2014
@@ -244,11 +244,16 @@
/* if no portnum specified, set default for transport */
if (!portnum) {
+ reg->portconfigured = FALSE;
if (transport == SIP_TRANSPORT_TLS) {
portnum = STANDARD_TLS_PORT;
} else {
portnum = STANDARD_SIP_PORT;
}
+ } else {
+ /* If any port (including 5060/5061) was configured we should avoid
+ doing SRV lookups - just like for SIP URI's */
+ reg->portconfigured = TRUE;
}
/* copy into sip_registry object */
Modified: team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/include/sip.h?view=diff&rev=421664&r1=421663&r2=421664
==============================================================================
--- team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/include/sip.h (original)
+++ team/oej/pgtips-srv-and-outbound-stuff-1.8/channels/sip/include/sip.h Thu Aug 21 04:41:13 2014
@@ -1352,6 +1352,7 @@
);
enum sip_transport transport; /*!< Transport for this registration UDP, TCP or TLS */
int portno; /*!< Optional port override */
+ int portconfigured; /*!< If port is configured, disable SRV lookups. */
int regdomainport; /*!< Port override for domainport */
int expire; /*!< Sched ID of expiration */
int configured_expiry; /*!< Configured value to use for the Expires header */
Modified: team/oej/pgtips-srv-and-outbound-stuff-1.8/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/configs/sip.conf.sample?view=diff&rev=421664&r1=421663&r2=421664
==============================================================================
--- team/oej/pgtips-srv-and-outbound-stuff-1.8/configs/sip.conf.sample (original)
+++ team/oej/pgtips-srv-and-outbound-stuff-1.8/configs/sip.conf.sample Thu Aug 21 04:41:13 2014
@@ -704,6 +704,7 @@
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
; they are blank. See the third example below for an illustration.
;
+; If a port number is specified for the host, no SRV lookup will be done.
;
; Examples:
;
More information about the asterisk-commits
mailing list