[asterisk-commits] oej: trunk r153904 - in /trunk: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 3 09:16:34 CST 2008
Author: oej
Date: Mon Nov 3 09:16:33 2008
New Revision: 153904
URL: http://svn.digium.com/view/asterisk?view=rev&rev=153904
Log:
Adding a separation of remote authentication and our authentication.
remotesecret => our password for a remote service
secret => our authentication when someone calls us
Secret => still has both functions if remotesecret is not used.
Modified:
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=153904&r1=153903&r2=153904
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Mon Nov 3 09:16:33 2008
@@ -11,6 +11,10 @@
making the new/old message count available to local devices.
* Added support for setting the domain in the URI for caller of an
outbound call by using the SIPFROMDOMAIN channel variable.
+ * Added a new configuration option "remotesecret" for authentication to
+ remote services. For backwards compatibility, "secret" still has the
+ same function as before, but now you can configure both a remote secret and a
+ local secret for mutual authentication.
Skinny Changes
--------------
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=153904&r1=153903&r2=153904
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Nov 3 09:16:33 2008
@@ -1751,14 +1751,17 @@
AST_LIST_ENTRY(sip_mailbox) entry;
};
-/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
+/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host)
+ \note This structure needs stringfields! Please!
+*/
/* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
struct sip_peer {
char name[80]; /*!< peer->name is the unique name of this object */
struct sip_socket socket; /*!< Socket used for this peer */
unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
- char secret[80]; /*!< Password */
+ char secret[80]; /*!< Password for inbound auth */
char md5secret[80]; /*!< Password in MD5 */
+ char remotesecret[80]; /*!< Remote secret (trunks, remote devices) */
struct sip_auth *auth; /*!< Realm authentication list */
char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
@@ -10420,8 +10423,9 @@
ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Registration is outgoing call */
r->call = dialog_ref(p, "copying dialog into registry r->call"); /* Save pointer to SIP dialog */
p->registry = registry_addref(r, "transmit_register: addref to p->registry in transmit_register"); /* Add pointer to registry in packet */
- if (!ast_strlen_zero(r->secret)) /* Secret (password) */
+ if (!ast_strlen_zero(r->secret)) { /* Secret (password) */
ast_string_field_set(p, peersecret, r->secret);
+ }
if (!ast_strlen_zero(r->md5secret))
ast_string_field_set(p, peermd5secret, r->md5secret);
/* User name in this realm
@@ -13899,6 +13903,7 @@
}
ast_cli(fd, " Secret : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>");
ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>");
+ ast_cli(fd, " Remote Secret: %s\n", ast_strlen_zero(peer->remotesecret)?"<Not set>":"<Set>");
for (auth = peer->auth; auth; auth = auth->next) {
ast_cli(fd, " Realm-auth : Realm %-15.15s User %-10.20s ", auth->realm, auth->username);
ast_cli(fd, "%s\n", !ast_strlen_zero(auth->secret)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>"));
@@ -14011,6 +14016,7 @@
astman_append(s, "ObjectName: %s\r\n", peer->name);
astman_append(s, "ChanObjectType: peer\r\n");
astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
+ astman_append(s, "RemoteSecretExist: %s\r\n", ast_strlen_zero(peer->remotesecret)?"N":"Y");
astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
astman_append(s, "Context: %s\r\n", peer->context);
astman_append(s, "Language: %s\r\n", peer->language);
@@ -21695,6 +21701,7 @@
peer->call_limit=999;
strcpy(peer->vmexten, default_vmexten);
peer->secret[0] = '\0';
+ peer->remotesecret[0] = '\0';
peer->md5secret[0] = '\0';
peer->cid_num[0] = '\0';
peer->cid_name[0] = '\0';
@@ -21866,9 +21873,11 @@
} else if (!strcasecmp(v->name, "type")) {
if (!strcasecmp(v->value, "peer"))
peer->onlymatchonip = TRUE; /* For realtime support, add type=peer in the table */
- } else if (!strcasecmp(v->name, "secret"))
+ } else if (!strcasecmp(v->name, "remotesecret")) {
+ ast_copy_string(peer->remotesecret, v->value, sizeof(peer->remotesecret));
+ } else if (!strcasecmp(v->name, "secret")) {
ast_copy_string(peer->secret, v->value, sizeof(peer->secret));
- else if (!strcasecmp(v->name, "md5secret"))
+ } else if (!strcasecmp(v->name, "md5secret"))
ast_copy_string(peer->md5secret, v->value, sizeof(peer->md5secret));
else if (!strcasecmp(v->name, "auth"))
peer->auth = add_realm_authentication(peer->auth, v->value, v->lineno);
@@ -22205,7 +22214,7 @@
if (!ast_strlen_zero(callback)) { /* build string from peer info */
char *reg_string;
- if (asprintf(®_string, "%s:%s@%s/%s", peer->username, peer->secret, peer->tohost, callback) < 0) {
+ if (asprintf(®_string, "%s:%s@%s/%s", peer->username, peer->remotesecret ? peer->remotesecret : peer->secret, peer->tohost, callback) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
} else if (reg_string) {
sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=153904&r1=153903&r2=153904
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Nov 3 09:16:33 2008
@@ -712,6 +712,7 @@
; callingpres callingpres
; permit permit
; deny deny
+; remotesecret
; secret secret
; md5secret md5secret
; transport transport
@@ -782,7 +783,7 @@
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
-;secret=guessit
+;remotesecret=guessit ; Our password to their service
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
@@ -802,7 +803,8 @@
;type=peer
;host=sip.provider1.com
;fromuser=4015552299 ; how your provider knows you
-;secret=youwillneverguessit
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=gissadetdu ; The password they use to contact us
;callbackextension=123 ; Register with this server and require calls coming back to this extension
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
; ; accept both tcp and udp. Default is udp. The first transport
More information about the asterisk-commits
mailing list