Merged revisions 221432 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
  r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines
  
  Merged revisions 221360 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
    
    Fix SRV lookup and Request-URI generation in chan_sip.
    
    This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct.  That field is used during RURI generation to determine if the port should be included in the RURI.  It is also used in some places to determine if an SRV lookup should occur.
    
    (closes issue #14418)
    Reported by: klaus3000
    Tested by: klaus3000, mnicholson
    
    Review: https://reviewboard.asterisk.org/r/369/
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221486 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 23 insertions, 3 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 23fee3496..d36bf8a98 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1233,6 +1233,7 @@ struct sip_pvt {
 	char tag[11];				/*!< Our tag for this session */
 	int sessionid;				/*!< SDP Session ID */
 	int sessionversion;			/*!< SDP Session Version */
+	int portinuri:1;			/*!< Non zero if a port has been specified, will also disable srv lookups */
 	int64_t sessionversion_remote;		/*!< Remote UA's SDP Session Version */
 	int session_modify;			/*!< Session modification request true/false  */
 	struct sockaddr_in sa;			/*!< Our peer */
@@ -1500,6 +1501,7 @@ struct sip_peer {
 	struct ast_dnsmgr_entry *dnsmgr;/*!<  DNS refresh manager for peer */
 	struct sockaddr_in addr;	/*!<  IP address of peer */
 	int maxcallbitrate;		/*!< Maximum Bitrate for a video call */
+	int portinuri:1;		/*!< Whether the port should be included in the URI */
 
 	/* Qualification */
 	struct sip_pvt *call;		/*!<  Call pointer */
@@ -4345,6 +4347,8 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
 	dialog->rtptimeout = peer->rtptimeout;
 	dialog->peerauth = peer->auth;
 	dialog->maxcallbitrate = peer->maxcallbitrate;
+	if (!dialog->portinuri)
+		dialog->portinuri = peer->portinuri;
 	if (ast_strlen_zero(dialog->tohost))
 		ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
 	if (!ast_strlen_zero(peer->fromdomain)) {
@@ -4406,8 +4410,10 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer, int newdialog)
 
 	ast_copy_string(peername, opeer, sizeof(peername));
 	port = strchr(peername, ':');
-	if (port)
+	if (port) {
 		*port++ = '\0';
+		dialog->portinuri = 1;
+	}
 	dialog->sa.sin_family = AF_INET;
 	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) */
@@ -9334,7 +9340,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
 			ast_str_append(&invite, 0, "%s@", n);
 		}
 		ast_str_append(&invite, 0, "%s", p->tohost);
-		if (ntohs(p->sa.sin_port) != STANDARD_SIP_PORT)
+		if (p->portinuri)
 			ast_str_append(&invite, 0, ":%d", ntohs(p->sa.sin_port));
 		ast_str_append(&invite, 0, "%s", urioptions);
 	}
@@ -10335,6 +10341,7 @@ static int expire_register(const void *data)
 	manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
 	register_peer_exten(peer, FALSE);	/* Remove regexten */
 	peer->expire = -1;
+	peer->portinuri = 0;
 	ast_device_state_changed("SIP/%s", peer->name);
 
 	/* Do we need to release this peer from memory? 
@@ -10556,6 +10563,7 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
 		peer->useragent[0] = '\0';
 		peer->sipoptions = 0;
 		peer->lastms = 0;
+		peer->portinuri = 0;
 		pvt->expiry = 0;
 
 		ast_verb(3, "Unregistered SIP '%s'\n", peer->name);
@@ -10574,6 +10582,11 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
 		ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
 	}
 
+	if (!ast_strlen_zero(pt))
+		peer->portinuri = 1;
+	else
+		peer->portinuri = 0;
+
 	/* handle the transport type specified in Contact header. */
 	if ((transport_type = get_transport_str2enum(transport))) {
 		/* if the port is not specified but the transport is, make sure to set the
@@ -21337,6 +21350,9 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 		/* XXX should unregister ? */
 	}
 
+	if (found)
+		peer->portinuri = 0;
+
 	/* If we have realm authentication information, remove them (reload) */
 	clear_realm_authentication(peer->auth);
 	peer->auth = NULL;
@@ -21471,6 +21487,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 					ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
 				}
 			} else if (!strcasecmp(v->name, "port")) {
+				peer->portinuri = 1;
 				if (!realtime && peer->host_dynamic) {
 					peer->defaddr.sin_port = htons(atoi(v->value));
 				} else {
@@ -21773,7 +21790,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 
 		if (srvlookup) {
 			if (ast_get_ip_or_srv(&peer->addr, srvlookup, 
-								global_srvlookup ?  
+								global_srvlookup && !peer->portinuri ?
 									((peer->socket.type & SIP_TRANSPORT_UDP) ? "_sip._udp" :
 									 (peer->socket.type & SIP_TRANSPORT_TCP) ? "_sip._tcp" :
 									 "_sip._tls")
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 05b4fca96..14b0f3976 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -146,6 +146,9 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; Disabling DNS SRV lookups disables the 
                                 ; ability to place SIP calls based on domain 
                                 ; names to some other SIP users on the Internet
+                                ; Specifying a port in a SIP peer definition or
+                                ; when dialing outbound calls will supress SRV
+                                ; lookups for that peer or call.
                                 
 ;pedantic=yes                   ; Enable checking of tags in headers, 
                                 ; international character conversions in URIs