[asterisk-commits] mnicholson: trunk r221432 - in /trunk: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 30 15:40:25 CDT 2009
Author: mnicholson
Date: Wed Sep 30 15:40:20 2009
New Revision: 221432
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=221432
Log:
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/
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=221432&r1=221431&r2=221432
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Sep 30 15:40:20 2009
@@ -1776,6 +1776,7 @@
long branch; /*!< The branch identifier of this session */
long invite_branch; /*!< The branch used when we sent the initial INVITE */
int64_t sessionversion_remote; /*!< Remote UA's SDP Session Version */
+ int portinuri:1; /*!< Non zero if a port has been specified, will also disable srv lookups */
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
@@ -2029,6 +2030,7 @@
struct sip_proxy *outboundproxy; /*!< Outbound proxy for this peer */
struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
struct sockaddr_in addr; /*!< IP address of peer */
+ int portinuri:1; /*!< Whether the port should be included in the URI */
struct sip_pvt *call; /*!< Call pointer */
int pokeexpire; /*!< Qualification: When to expire poke (qualify= checking) */
int lastms; /*!< Qualification: How long last response took (in ms), or -1 for no response */
@@ -3209,11 +3211,15 @@
* pt buffer is provided or the pt has errors when being converted
* to an int value, the port provided as the standard is used.
*/
-static int port_str2int(const char *pt, unsigned int standard)
+static int port_str2int(const char *pt, unsigned int standard, int *found_port)
{
int port = standard;
if (ast_strlen_zero(pt) || (sscanf(pt, "%30d", &port) != 1) || (port < 1) || (port > 65535)) {
port = standard;
+ if (found_port)
+ *found_port = 0;
+ } else if (found_port) {
+ *found_port = 1;
}
return port;
@@ -3233,7 +3239,7 @@
return NULL;
proxy->force = force;
ast_copy_string(proxy->name, name, sizeof(proxy->name));
- proxy->ip.sin_port = htons(port_str2int(port, STANDARD_SIP_PORT));
+ proxy->ip.sin_port = htons(port_str2int(port, STANDARD_SIP_PORT, NULL));
proxy_update(proxy);
return proxy;
}
@@ -5277,6 +5283,8 @@
dialog->noncodeccapability &= ~AST_RTP_DTMF;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
+ if (!dialog->portinuri)
+ dialog->portinuri = peer->portinuri;
dialog->chanvars = copy_vars(peer->chanvars);
@@ -5299,8 +5307,10 @@
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) */
@@ -5335,7 +5345,7 @@
/* This address should be updated using dnsmgr */
memcpy(&dialog->sa.sin_addr, &sin->sin_addr, sizeof(dialog->sa.sin_addr));
if (!sin->sin_port) {
- portno = port_str2int(port, (dialog->socket.type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT);
+ portno = port_str2int(port, (dialog->socket.type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT, NULL);
} else {
portno = ntohs(sin->sin_port);
}
@@ -5362,7 +5372,7 @@
}
}
if (!portno)
- portno = port_str2int(port, (dialog->socket.type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT);
+ portno = port_str2int(port, (dialog->socket.type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT, NULL);
hp = ast_gethostbyname(hostn, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "No such host: %s\n", peername);
@@ -10804,7 +10814,7 @@
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);
}
@@ -12192,6 +12202,7 @@
return 0;
peer->expire = -1;
+ peer->portinuri = 0;
memset(&peer->addr, 0, sizeof(peer->addr));
destroy_association(peer); /* remove registration data from storage */
@@ -12343,9 +12354,9 @@
/* set port */
if (((get_transport_str2enum(transport) == SIP_TRANSPORT_TLS)) || !(strncasecmp(fullcontact, "sips", 4))) {
- port = port_str2int(pt, STANDARD_TLS_PORT);
+ port = port_str2int(pt, STANDARD_TLS_PORT, NULL);
} else {
- port = port_str2int(pt, STANDARD_SIP_PORT);
+ port = port_str2int(pt, STANDARD_SIP_PORT, NULL);
}
@@ -12390,6 +12401,7 @@
int expire = atoi(expires);
char *curi, *host, *pt, *transport;
int port;
+ int portinuri;
int transport_type;
const char *useragent;
struct hostent *hp;
@@ -12443,6 +12455,7 @@
ast_string_field_set(peer, useragent, "");
peer->sipoptions = 0;
peer->lastms = 0;
+ peer->portinuri = 0;
pvt->expiry = 0;
ast_verb(3, "Unregistered SIP '%s'\n", peer->name);
@@ -12468,11 +12481,12 @@
* default port to match the specified transport. This may or may not be the
* same transport used by the pvt struct for the Register dialog. */
- port = port_str2int(pt, (transport_type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT);
+ port = port_str2int(pt, (transport_type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT, &portinuri);
} else {
- port = port_str2int(pt, STANDARD_SIP_PORT);
+ port = port_str2int(pt, STANDARD_SIP_PORT, &portinuri);
transport_type = pvt->socket.type;
}
+ peer->portinuri = portinuri;
/* if the peer's socket type is different than the Registration
* transport type, change it. If it got this far, it is a
@@ -14145,7 +14159,7 @@
if (pt)
*pt++ = '\0'; /* remember port pointer */
p->sa = p->recv;
- p->sa.sin_port = htons(port_str2int(pt, STANDARD_SIP_PORT));
+ p->sa.sin_port = htons(port_str2int(pt, STANDARD_SIP_PORT, NULL));
if (sip_debug_test_pvt(p)) {
const struct sockaddr_in *dst = sip_real_dst(p);
@@ -24214,6 +24228,9 @@
/* 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;
@@ -24365,6 +24382,7 @@
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 {
@@ -24620,7 +24638,7 @@
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(peer->socket.type));
- if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup ? transport : NULL)) {
+ if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup && !peer->portinuri ? transport : NULL)) {
ast_log(LOG_ERROR, "srvlookup failed for host: %s, on peer %s, removing peer\n", _srvlookup, peer->name);
unref_peer(peer, "getting rid of a peer pointer");
return NULL;
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=221432&r1=221431&r2=221432
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Wed Sep 30 15:40:20 2009
@@ -152,6 +152,9 @@
; 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
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