[asterisk-commits] bbryant: branch 1.6.0 r127163 - in /branches/1.6.0: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 1 16:20:00 CDT 2008
Author: bbryant
Date: Tue Jul 1 16:19:59 2008
New Revision: 127163
URL: http://svn.digium.com/view/asterisk?view=rev&rev=127163
Log:
Merged revisions 127154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines
Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user.
........
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/channels/chan_sip.c
branches/1.6.0/configs/sip.conf.sample
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=127163&r1=127162&r2=127163
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Jul 1 16:19:59 2008
@@ -410,6 +410,12 @@
SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
};
+/*!< Define some SIP transports */
+enum sip_transport {
+ SIP_TRANSPORT_UDP = 1,
+ SIP_TRANSPORT_TCP = 1 << 1,
+ SIP_TRANSPORT_TLS = 1 << 2,
+};
/*! \brief definition of a sip proxy server
*
@@ -421,6 +427,7 @@
char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
struct sockaddr_in ip; /*!< Currently used IP address and port */
time_t last_dnsupdate; /*!< When this was resolved */
+ enum sip_transport transport;
int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
/* Room for a SRV record chain based on the name */
};
@@ -766,13 +773,6 @@
#define INC_CALL_LIMIT 1
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
-
-/*!< Define some SIP transports */
-enum sip_transport {
- SIP_TRANSPORT_UDP = 1,
- SIP_TRANSPORT_TCP = 1 << 1,
- SIP_TRANSPORT_TLS = 1 << 2,
-};
/*!< The SIP socket definition */
struct sip_socket {
@@ -1708,6 +1708,7 @@
static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin);
static int sip_standard_port(struct sip_socket s);
static int sip_prepare_socket(struct sip_pvt *p);
+static int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
/*--- Transmitting responses and requests */
static int sipsock_read(int *id, int fd, short events, void *ignore);
@@ -2521,6 +2522,14 @@
return "UNKNOWN";
}
+static inline const char *get_transport_pvt(struct sip_pvt *p)
+{
+ if (p->outboundproxy && p->outboundproxy->transport)
+ p->socket.type = p->outboundproxy->transport;
+
+ return get_transport(p->socket.type);
+}
+
/*! \brief Transmit SIP message
Sends a SIP request or response on a given socket (in the pvt)
Called by retrans_pkt, send_request, send_response and
@@ -2531,7 +2540,7 @@
int res = 0;
const struct sockaddr_in *dst = sip_real_dst(p);
- ast_debug(1, "Trying to put '%.10s' onto %s socket...\n", data, get_transport(p->socket.type));
+ ast_debug(1, "Trying to put '%.10s' onto %s socket destined for %s:%d\n", data, get_transport_pvt(p), ast_inet_ntoa(dst->sin_addr), htons(dst->sin_port));
if (sip_prepare_socket(p) < 0)
return XMIT_ERROR;
@@ -2575,7 +2584,7 @@
/* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
ast_string_field_build(p, via, "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x%s",
- get_transport(p->socket.type),
+ get_transport_pvt(p),
ast_inet_ntoa(p->ourip.sin_addr),
ntohs(p->ourip.sin_port), p->branch, rport);
}
@@ -6018,35 +6027,14 @@
char buf[256] = "";
char *username = NULL;
char *hostname=NULL, *secret=NULL, *authuser=NULL;
- char *porta=NULL;
char *callback=NULL;
- char *trans=NULL;
if (!value)
return -1;
ast_copy_string(buf, value, sizeof(buf));
- username = strstr(buf, "://");
-
- if (username) {
- *username = '\0';
- username += 3;
-
- trans = buf;
-
- if (!strcasecmp(trans, "udp"))
- transport = SIP_TRANSPORT_UDP;
- else if (!strcasecmp(trans, "tcp"))
- transport = SIP_TRANSPORT_TCP;
- else if (!strcasecmp(trans, "tls"))
- transport = SIP_TRANSPORT_TLS;
- else
- ast_log(LOG_WARNING, "'%s' is not a valid transport value for registration '%s' at line '%d'\n", trans, value, lineno);
- } else {
- username = buf;
- ast_debug(1, "no trans\n");
- }
+ sip_parse_host(buf, lineno, &username, &portnum, &transport);
/* First split around the last '@' then parse the two components. */
hostname = strrchr(username, '@'); /* allow @ in the first part */
@@ -6070,18 +6058,6 @@
*callback++ = '\0';
if (ast_strlen_zero(callback))
callback = "s";
- porta = strchr(hostname, ':');
- if (porta) {
- *porta++ = '\0';
- portnum = atoi(porta);
- if (portnum == 0) {
- ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
- return -1;
- }
- } else {
- portnum = (transport == SIP_TRANSPORT_TLS) ?
- STANDARD_TLS_PORT : STANDARD_SIP_PORT;
- }
if (!(reg = ast_calloc(1, sizeof(*reg)))) {
ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
return -1;
@@ -8515,7 +8491,7 @@
else
ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr));
} else
- ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d;transport=%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ntohs(p->socket.port), get_transport(p->socket.type));
+ ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d;transport=%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ntohs(p->socket.port), get_transport_pvt(p));
}
/*! \brief Build the Remote Party-ID & From using callingpres options */
@@ -18288,6 +18264,10 @@
if (s->fd != -1)
return s->fd;
+ if (p->outboundproxy && p->outboundproxy->transport) {
+ s->type = p->outboundproxy->transport;
+ }
+
if (s->type & SIP_TRANSPORT_UDP) {
s->fd = sipsock;
return s->fd;
@@ -18344,6 +18324,50 @@
}
return s->fd;
+}
+
+/*!
+ * \brief Small function to parse a config line for a host with a transport
+ * i.e. tls://www.google.com:8056
+ */
+static int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport)
+{
+ char *port;
+
+ if ((*hostname = strstr(line, "://"))) {
+ *hostname += 3;
+
+ if (!strncasecmp(line, "tcp", 3))
+ *transport = SIP_TRANSPORT_TCP;
+ else if (!strncasecmp(line, "tls", 3))
+ *transport = SIP_TRANSPORT_TLS;
+ else if (!strncasecmp(line, "udp", 3))
+ *transport = SIP_TRANSPORT_UDP;
+ else
+ ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", line, lineno);
+ } else {
+ *hostname = line;
+ *transport = SIP_TRANSPORT_UDP;
+ }
+
+ if ((port = strchr(*hostname, ':'))) {
+ *port++ = '\0';
+
+ if (!sscanf(port, "%u", portnum)) {
+ ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", port, lineno);
+ port = NULL;
+ }
+ }
+
+ if (!port) {
+ if (*transport & SIP_TRANSPORT_TLS) {
+ *portnum = STANDARD_TLS_PORT;
+ } else {
+ *portnum = STANDARD_SIP_PORT;
+ }
+ }
+
+ return 0;
}
/*!
@@ -20548,21 +20572,33 @@
} else if (!strcasecmp(v->name, "fromdomain")) {
ast_copy_string(default_fromdomain, v->value, sizeof(default_fromdomain));
} else if (!strcasecmp(v->name, "outboundproxy")) {
- char *name, *port = NULL, *force;
-
- name = ast_strdupa(v->value);
- if ((port = strchr(name, ':'))) {
- *port++ = '\0';
- global_outboundproxy.ip.sin_port = htons(atoi(port));
+ int portnum;
+ char *tok, *proxyname;
+
+ if (ast_strlen_zero(v->value)) {
+ ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf.", v->lineno);
+ continue;
}
- if ((force = strchr(port ? port : name, ','))) {
- *force++ = '\0';
- global_outboundproxy.force = (!strcasecmp(force, "force"));
+ tok = ast_skip_blanks(strtok(ast_strdupa(v->value), ","));
+
+ sip_parse_host(tok, v->lineno, &proxyname, &portnum, &global_outboundproxy.transport);
+
+ global_outboundproxy.ip.sin_port = htons(portnum);
+
+ if ((tok = strtok(NULL, ","))) {
+ global_outboundproxy.force = !strncasecmp(ast_skip_blanks(tok), "force", 5);
+ } else {
+ global_outboundproxy.force = FALSE;
}
- ast_copy_string(global_outboundproxy.name, name, sizeof(global_outboundproxy.name));
- proxy_update(&global_outboundproxy);
-
+
+ if (ast_strlen_zero(proxyname)) {
+ ast_log(LOG_WARNING, "you must specify a name for the outboundproxy on line %d of sip.conf.", v->lineno);
+ global_outboundproxy.name[0] = '\0';
+ continue;
+ }
+
+ ast_copy_string(global_outboundproxy.name, proxyname, sizeof(global_outboundproxy.name));
} else if (!strcasecmp(v->name, "autocreatepeer")) {
autocreatepeer = ast_true(v->value);
} else if (!strcasecmp(v->name, "match_auth_username")) {
Modified: branches/1.6.0/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/sip.conf.sample?view=diff&rev=127163&r1=127162&r2=127163
==============================================================================
--- branches/1.6.0/configs/sip.conf.sample (original)
+++ branches/1.6.0/configs/sip.conf.sample Tue Jul 1 16:19:59 2008
@@ -233,6 +233,9 @@
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
+;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
+; ; (could also be tcp,udp) - defining transports on the proxy line only
+; ; applies for the global proxy, otherwise use the transport= option
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
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