[asterisk-commits] rizzo: trunk r45945 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Oct 23 04:08:49 MST 2006
Author: rizzo
Date: Mon Oct 23 06:08:47 2006
New Revision: 45945
URL: http://svn.digium.com/view/asterisk?rev=45945&view=rev
Log:
More cleanup of check_user_full with no functional change
apart from a small (but disabled by default) new option.
In detail:
+ introduce a new value for enum check_auth_result, AUTH_DONT_KNOW,
used (read below) when a function does not have a conclusive response.
Possibly this is the same as AUTH_NOT_FOUND, but need to check further.
+ move the large blocks (checking in the users list and in the peers
list, respectively) from check_user_full() to separate functions.
They return AUTH_DONT_KNOW in case they don't find a match, so
the caller know that it has to try the next method.
There is still some duplication of code here, but i
have not tried yet to remove it.
+ [new option] a new option in sip.conf, match_auth_username,
has been introduced, and disabled by default.
If set, and the incoming request carries authentication info,
the username to match in the users list is taken from there
rather than from the From: field.
This change is easy to identify, being made of
- one line to declare the variable match_auth_username
- a block of 15 lines in check_user_full()
- one line in sip list settings
- two lines for parsing the config file.
check_user_full() is now a lot cleaner - basically a sequence of
checks that are applied to the request. This will help future
work with new matching schemes.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=45945&r1=45944&r2=45945&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Oct 23 06:08:47 2006
@@ -317,6 +317,9 @@
/*! \brief Authentication result from check_auth* functions */
enum check_auth_result {
+ AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
+ /* XXX maybe this is the same as AUTH_NOT_FOUND */
+
AUTH_SUCCESSFUL = 0,
AUTH_CHALLENGE_SENT = 1,
AUTH_SECRET_FAILED = -1,
@@ -516,6 +519,7 @@
static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
static int pedanticsipchecking; /*!< Extra checking ? Default off */
static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
+static int match_auth_username; /*!< Match auth username if available instead of From: Default off. */
static int global_relaxdtmf; /*!< Relax DTMF */
static int global_rtptimeout; /*!< Time out call if no RTP */
static int global_rtpholdtimeout;
@@ -8852,6 +8856,247 @@
return res;
}
+static enum check_auth_result check_user_ok(struct sip_pvt *p, char *of,
+ struct sip_request *req, int sipmethod, struct sockaddr_in *sin,
+ enum xmittype reliable,
+ char *rpid_num, char *calleridname, char *uri2)
+{
+ enum check_auth_result res;
+ struct sip_user *user = find_user(of, 1);
+ int debug=sip_debug_test_addr(sin);
+
+ /* Find user based on user name in the from header */
+ if (!user) {
+ if (debug)
+ ast_verbose("No user '%s' in SIP users list\n", of);
+ return AUTH_DONT_KNOW;
+ }
+ if (!ast_apply_ha(user->ha, sin)) {
+ if (debug)
+ ast_verbose("Found user '%s' for '%s', but fails host access\n",
+ user->name, of);
+ ASTOBJ_UNREF(user,sip_destroy_user);
+ return AUTH_DONT_KNOW;
+ }
+ if (debug)
+ ast_verbose("Found user '%s' for '%s'\n", user->name, of);
+
+ ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
+ ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
+ /* copy channel vars */
+ p->chanvars = copy_vars(user->chanvars);
+ p->prefs = user->prefs;
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &p->prefs);
+ p->autoframing = user->autoframing;
+ }
+ /* replace callerid if rpid found, and not restricted */
+ if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
+ char *tmp;
+ if (*calleridname)
+ ast_string_field_set(p, cid_name, calleridname);
+ tmp = ast_strdupa(rpid_num);
+ if (ast_is_shrinkable_phonenumber(tmp))
+ ast_shrink_phone_number(tmp);
+ ast_string_field_set(p, cid_num, tmp);
+ }
+
+ do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE) );
+
+ if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri2, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) {
+ sip_cancel_destroy(p);
+ ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
+ ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
+ /* Copy SIP extensions profile from INVITE */
+ if (p->sipoptions)
+ user->sipoptions = p->sipoptions;
+
+ /* If we have a call limit, set flag */
+ if (user->call_limit)
+ ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
+ if (!ast_strlen_zero(user->context))
+ ast_string_field_set(p, context, user->context);
+ if (!ast_strlen_zero(user->cid_num) && !ast_strlen_zero(p->cid_num)) {
+ char *tmp = ast_strdupa(user->cid_num);
+ if (ast_is_shrinkable_phonenumber(tmp))
+ ast_shrink_phone_number(tmp);
+ ast_string_field_set(p, cid_num, tmp);
+ }
+ if (!ast_strlen_zero(user->cid_name) && !ast_strlen_zero(p->cid_num))
+ ast_string_field_set(p, cid_name, user->cid_name);
+ ast_string_field_set(p, username, user->name);
+ ast_string_field_set(p, peername, user->name);
+ ast_string_field_set(p, peersecret, user->secret);
+ ast_string_field_set(p, peermd5secret, user->md5secret);
+ ast_string_field_set(p, subscribecontext, user->subscribecontext);
+ ast_string_field_set(p, accountcode, user->accountcode);
+ ast_string_field_set(p, language, user->language);
+ ast_string_field_set(p, mohsuggest, user->mohsuggest);
+ ast_string_field_set(p, mohinterpret, user->mohinterpret);
+ p->allowtransfer = user->allowtransfer;
+ p->amaflags = user->amaflags;
+ p->callgroup = user->callgroup;
+ p->pickupgroup = user->pickupgroup;
+ if (user->callingpres) /* User callingpres setting will override RPID header */
+ p->callingpres = user->callingpres;
+
+ /* Set default codec settings for this call */
+ p->capability = user->capability; /* User codec choice */
+ p->jointcapability = user->capability; /* Our codecs */
+ if (p->peercapability) /* AND with peer's codecs */
+ p->jointcapability &= p->peercapability;
+ if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
+ (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
+ p->noncodeccapability |= AST_RTP_DTMF;
+ else
+ p->noncodeccapability &= ~AST_RTP_DTMF;
+ if (p->t38.peercapability)
+ p->t38.jointcapability &= p->t38.peercapability;
+ p->maxcallbitrate = user->maxcallbitrate;
+ /* If we do not support video, remove video from call structure */
+ if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
+ ast_rtp_destroy(p->vrtp);
+ p->vrtp = NULL;
+ }
+ }
+ ASTOBJ_UNREF(user, sip_destroy_user);
+ return res;
+}
+
+static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
+ struct sip_request *req, int sipmethod, struct sockaddr_in *sin,
+ struct sip_peer **authpeer,
+ enum xmittype reliable,
+ char *rpid_num, char *calleridname, char *uri2)
+{
+ enum check_auth_result res;
+ int debug=sip_debug_test_addr(sin);
+ struct sip_peer *peer;
+
+ if (sipmethod == SIP_SUBSCRIBE)
+ /* For subscribes, match on peer name only */
+ peer = find_peer(of, NULL, 1);
+ else
+ /* Look for peer based on the IP address we received data from */
+ /* If peer is registered from this IP address or have this as a default
+ IP address, this call is from the peer
+ */
+ peer = find_peer(NULL, &p->recv, 1);
+
+ if (!peer) {
+ if (debug)
+ ast_verbose("No matching peer for '%s' from '%s:%d'\n",
+ of, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
+ return AUTH_DONT_KNOW;
+ }
+
+ if (debug)
+ ast_verbose("Found peer '%s' for '%s' from %s:%d\n",
+ peer->name, of, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
+
+ /* XXX what about p->prefs = peer->prefs; ? */
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ }
+
+ /* Take the peer */
+ ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
+ ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
+
+ /* Copy SIP extensions profile to peer */
+ if (p->sipoptions)
+ peer->sipoptions = p->sipoptions;
+
+ /* replace callerid if rpid found, and not restricted */
+ if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
+ char *tmp = ast_strdupa(rpid_num);
+ if (*calleridname)
+ ast_string_field_set(p, cid_name, calleridname);
+ if (ast_is_shrinkable_phonenumber(tmp))
+ ast_shrink_phone_number(tmp);
+ ast_string_field_set(p, cid_num, tmp);
+ }
+ do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE));
+
+ ast_string_field_set(p, peersecret, peer->secret);
+ ast_string_field_set(p, peermd5secret, peer->md5secret);
+ ast_string_field_set(p, subscribecontext, peer->subscribecontext);
+ ast_string_field_set(p, mohinterpret, peer->mohinterpret);
+ ast_string_field_set(p, mohsuggest, peer->mohsuggest);
+ if (peer->callingpres) /* Peer calling pres setting will override RPID */
+ p->callingpres = peer->callingpres;
+ if (peer->maxms && peer->lastms)
+ p->timer_t1 = peer->lastms;
+ if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
+ /* Pretend there is no required authentication */
+ ast_string_field_free(p, peersecret);
+ ast_string_field_free(p, peermd5secret);
+ }
+ if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) {
+ ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
+ ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
+ /* If we have a call limit, set flag */
+ if (peer->call_limit)
+ ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
+ ast_string_field_set(p, peername, peer->name);
+ ast_string_field_set(p, authname, peer->name);
+
+ /* copy channel vars */
+ p->chanvars = copy_vars(peer->chanvars);
+ if (authpeer) {
+ (*authpeer) = ASTOBJ_REF(peer); /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
+ }
+
+ if (!ast_strlen_zero(peer->username)) {
+ ast_string_field_set(p, username, peer->username);
+ /* Use the default username for authentication on outbound calls */
+ /* XXX this takes the name from the caller... can we override ? */
+ ast_string_field_set(p, authname, peer->username);
+ }
+ if (!ast_strlen_zero(peer->cid_num) && !ast_strlen_zero(p->cid_num)) {
+ char *tmp = ast_strdupa(peer->cid_num);
+ if (ast_is_shrinkable_phonenumber(tmp))
+ ast_shrink_phone_number(tmp);
+ ast_string_field_set(p, cid_num, tmp);
+ }
+ if (!ast_strlen_zero(peer->cid_name) && !ast_strlen_zero(p->cid_name))
+ ast_string_field_set(p, cid_name, peer->cid_name);
+ ast_string_field_set(p, fullcontact, peer->fullcontact);
+ if (!ast_strlen_zero(peer->context))
+ ast_string_field_set(p, context, peer->context);
+ ast_string_field_set(p, peersecret, peer->secret);
+ ast_string_field_set(p, peermd5secret, peer->md5secret);
+ ast_string_field_set(p, language, peer->language);
+ ast_string_field_set(p, accountcode, peer->accountcode);
+ p->amaflags = peer->amaflags;
+ p->callgroup = peer->callgroup;
+ p->pickupgroup = peer->pickupgroup;
+ p->capability = peer->capability;
+ p->prefs = peer->prefs;
+ p->jointcapability = peer->capability;
+ if (p->peercapability)
+ p->jointcapability &= p->peercapability;
+ p->maxcallbitrate = peer->maxcallbitrate;
+ if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
+ ast_rtp_destroy(p->vrtp);
+ p->vrtp = NULL;
+ }
+ if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
+ (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
+ p->noncodeccapability |= AST_RTP_DTMF;
+ else
+ p->noncodeccapability &= ~AST_RTP_DTMF;
+ if (p->t38.peercapability)
+ p->t38.jointcapability &= p->t38.peercapability;
+ }
+ ASTOBJ_UNREF(peer, sip_destroy_peer);
+ return res;
+}
+
+
/*! \brief Check if matching user or peer is defined
Match user on From: user name and peer on IP/port
This is used on first invite (not re-invites) and subscribe requests
@@ -8861,8 +9106,6 @@
int sipmethod, char *uri, enum xmittype reliable,
struct sockaddr_in *sin, struct sip_peer **authpeer)
{
- struct sip_user *user = NULL;
- struct sip_peer *peer;
char from[256], *c;
char *of;
char rpid_num[50];
@@ -8870,7 +9113,6 @@
enum check_auth_result res;
char *t;
char calleridname[50];
- int debug=sip_debug_test_addr(sin);
char *uri2 = ast_strdupa(uri);
/* Terminate URI */
@@ -8924,221 +9166,39 @@
if (ast_strlen_zero(of))
return AUTH_SUCCESSFUL;
- if (!authpeer) /* If we are looking for a peer, don't check the user objects (or realtime) */
- user = find_user(of, 1);
-
- /* Find user based on user name in the from header */
- if (user && !ast_apply_ha(user->ha, sin)) {
- if (!authpeer && debug)
- ast_verbose("Found user '%s', but fails host access\n", user->name);
- ASTOBJ_UNREF(user,sip_destroy_user);
- user = NULL;
- }
- if (user) {
- ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- /* copy channel vars */
- p->chanvars = copy_vars(user->chanvars);
- p->prefs = user->prefs;
- /* Set Frame packetization */
- if (p->rtp) {
- ast_rtp_codec_setpref(p->rtp, &p->prefs);
- p->autoframing = user->autoframing;
- }
- /* replace callerid if rpid found, and not restricted */
- if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
- char *tmp;
- if (*calleridname)
- ast_string_field_set(p, cid_name, calleridname);
- tmp = ast_strdupa(rpid_num);
- if (ast_is_shrinkable_phonenumber(tmp))
- ast_shrink_phone_number(tmp);
- ast_string_field_set(p, cid_num, tmp);
- }
-
- do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE) );
-
- if (!(res = check_auth(p, req, user->name, user->secret, user->md5secret, sipmethod, uri2, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) {
- sip_cancel_destroy(p);
- ast_copy_flags(&p->flags[0], &user->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &user->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- /* Copy SIP extensions profile from INVITE */
- if (p->sipoptions)
- user->sipoptions = p->sipoptions;
-
- /* If we have a call limit, set flag */
- if (user->call_limit)
- ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
- if (!ast_strlen_zero(user->context))
- ast_string_field_set(p, context, user->context);
- if (!ast_strlen_zero(user->cid_num) && !ast_strlen_zero(p->cid_num)) {
- char *tmp = ast_strdupa(user->cid_num);
- if (ast_is_shrinkable_phonenumber(tmp))
- ast_shrink_phone_number(tmp);
- ast_string_field_set(p, cid_num, tmp);
- }
- if (!ast_strlen_zero(user->cid_name) && !ast_strlen_zero(p->cid_num))
- ast_string_field_set(p, cid_name, user->cid_name);
- ast_string_field_set(p, username, user->name);
- ast_string_field_set(p, peername, user->name);
- ast_string_field_set(p, peersecret, user->secret);
- ast_string_field_set(p, peermd5secret, user->md5secret);
- ast_string_field_set(p, subscribecontext, user->subscribecontext);
- ast_string_field_set(p, accountcode, user->accountcode);
- ast_string_field_set(p, language, user->language);
- ast_string_field_set(p, mohsuggest, user->mohsuggest);
- ast_string_field_set(p, mohinterpret, user->mohinterpret);
- p->allowtransfer = user->allowtransfer;
- p->amaflags = user->amaflags;
- p->callgroup = user->callgroup;
- p->pickupgroup = user->pickupgroup;
- if (user->callingpres) /* User callingpres setting will override RPID header */
- p->callingpres = user->callingpres;
-
- /* Set default codec settings for this call */
- p->capability = user->capability; /* User codec choice */
- p->jointcapability = user->capability; /* Our codecs */
- if (p->peercapability) /* AND with peer's codecs */
- p->jointcapability &= p->peercapability;
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
- p->noncodeccapability |= AST_RTP_DTMF;
- else
- p->noncodeccapability &= ~AST_RTP_DTMF;
- if (p->t38.peercapability)
- p->t38.jointcapability &= p->t38.peercapability;
- p->maxcallbitrate = user->maxcallbitrate;
- /* If we do not support video, remove video from call structure */
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
- ast_rtp_destroy(p->vrtp);
- p->vrtp = NULL;
- }
- }
- if (debug)
- ast_verbose("Found user '%s'\n", user->name);
- ASTOBJ_UNREF(user, sip_destroy_user);
+ if (match_auth_username) {
+ /*
+ * XXX This is experimental code to grab the search key from the
+ * Auth header's username instead of the 'From' name, if available.
+ * Do not enable this block unless you understand the side effects (if any!)
+ * Note, the search for "username" should be done in a more robust way.
+ * Note2, at the moment we chech both fields, though maybe we should
+ * pick one or another depending on the request ? XXX
+ */
+ const char *hdr = get_header(req, "Authorization");
+ if (ast_strlen_zero(hdr))
+ hdr = get_header(req, "Proxy-Authorization");
+
+ if ( !ast_strlen_zero(hdr) && (hdr = strstr(hdr, "username=\"")) ) {
+ ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
+ of = from;
+ of = strsep(&of, "\"");
+ }
+ }
+
+ if (!authpeer) {
+ /* If we are looking for a peer, don't check the
+ user objects (or realtime) */
+ res = check_user_ok(p, of, req, sipmethod, sin,
+ reliable, rpid_num, calleridname, uri2);
+ if (res != AUTH_DONT_KNOW)
+ return res;
+ }
+
+ res = check_peer_ok(p, of, req, sipmethod, sin,
+ authpeer, reliable, rpid_num, calleridname, uri2);
+ if (res != AUTH_DONT_KNOW)
return res;
- }
-
- /* XXX need to reindent the next block */
-
- /* If we didn't find a user match, check for peers */
- if (sipmethod == SIP_SUBSCRIBE)
- /* For subscribes, match on peer name only */
- peer = find_peer(of, NULL, 1);
- else
- /* Look for peer based on the IP address we received data from */
- /* If peer is registered from this IP address or have this as a default
- IP address, this call is from the peer
- */
- peer = find_peer(NULL, &p->recv, 1);
-
- if (!peer) {
- if (debug)
- ast_verbose("Found no matching peer or user for '%s:%d'\n", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
-
- } else {
- /* Set Frame packetization */
- if (p->rtp) {
- ast_rtp_codec_setpref(p->rtp, &peer->prefs);
- p->autoframing = peer->autoframing;
- }
- if (debug)
- ast_verbose("Found peer '%s'\n", peer->name);
-
- /* Take the peer */
- ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
-
- /* Copy SIP extensions profile to peer */
- if (p->sipoptions)
- peer->sipoptions = p->sipoptions;
-
- /* replace callerid if rpid found, and not restricted */
- if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
- char *tmp = ast_strdupa(rpid_num);
- if (*calleridname)
- ast_string_field_set(p, cid_name, calleridname);
- if (ast_is_shrinkable_phonenumber(tmp))
- ast_shrink_phone_number(tmp);
- ast_string_field_set(p, cid_num, tmp);
- }
- do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE));
-
- ast_string_field_set(p, peersecret, peer->secret);
- ast_string_field_set(p, peermd5secret, peer->md5secret);
- ast_string_field_set(p, subscribecontext, peer->subscribecontext);
- ast_string_field_set(p, mohinterpret, peer->mohinterpret);
- ast_string_field_set(p, mohsuggest, peer->mohsuggest);
- if (peer->callingpres) /* Peer calling pres setting will override RPID */
- p->callingpres = peer->callingpres;
- if (peer->maxms && peer->lastms)
- p->timer_t1 = peer->lastms;
- if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
- /* Pretend there is no required authentication */
- ast_string_field_free(p, peersecret);
- ast_string_field_free(p, peermd5secret);
- }
- if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable, ast_test_flag(req, SIP_PKT_IGNORE)))) {
- ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- /* If we have a call limit, set flag */
- if (peer->call_limit)
- ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
- ast_string_field_set(p, peername, peer->name);
- ast_string_field_set(p, authname, peer->name);
-
- /* copy channel vars */
- p->chanvars = copy_vars(peer->chanvars);
- if (authpeer) {
- (*authpeer) = ASTOBJ_REF(peer); /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
- }
-
- if (!ast_strlen_zero(peer->username)) {
- ast_string_field_set(p, username, peer->username);
- /* Use the default username for authentication on outbound calls */
- /* XXX this takes the name from the caller... can we override ? */
- ast_string_field_set(p, authname, peer->username);
- }
- if (!ast_strlen_zero(peer->cid_num) && !ast_strlen_zero(p->cid_num)) {
- char *tmp = ast_strdupa(peer->cid_num);
- if (ast_is_shrinkable_phonenumber(tmp))
- ast_shrink_phone_number(tmp);
- ast_string_field_set(p, cid_num, tmp);
- }
- if (!ast_strlen_zero(peer->cid_name) && !ast_strlen_zero(p->cid_name))
- ast_string_field_set(p, cid_name, peer->cid_name);
- ast_string_field_set(p, fullcontact, peer->fullcontact);
- if (!ast_strlen_zero(peer->context))
- ast_string_field_set(p, context, peer->context);
- ast_string_field_set(p, peersecret, peer->secret);
- ast_string_field_set(p, peermd5secret, peer->md5secret);
- ast_string_field_set(p, language, peer->language);
- ast_string_field_set(p, accountcode, peer->accountcode);
- p->amaflags = peer->amaflags;
- p->callgroup = peer->callgroup;
- p->pickupgroup = peer->pickupgroup;
- p->capability = peer->capability;
- p->prefs = peer->prefs;
- p->jointcapability = peer->capability;
- if (p->peercapability)
- p->jointcapability &= p->peercapability;
- p->maxcallbitrate = peer->maxcallbitrate;
- if (!ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) {
- ast_rtp_destroy(p->vrtp);
- p->vrtp = NULL;
- }
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
- p->noncodeccapability |= AST_RTP_DTMF;
- else
- p->noncodeccapability &= ~AST_RTP_DTMF;
- if (p->t38.peercapability)
- p->t38.jointcapability &= p->t38.peercapability;
- }
- ASTOBJ_UNREF(peer, sip_destroy_peer);
- return res;
- }
/* Finally, apply the guest policy */
if (global_allowguest)
@@ -10164,6 +10224,7 @@
ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
ast_cli(fd, " Videosupport: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "Yes" : "No");
ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No");
+ ast_cli(fd, " MatchAuthUsername: %s\n", match_auth_username ? "Yes" : "No");
ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No");
ast_cli(fd, " Allow subscriptions: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
ast_cli(fd, " Allow overlap dialing: %s\n", ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No");
@@ -15994,6 +16055,8 @@
outboundproxyip.sin_port = htons(format);
} else if (!strcasecmp(v->name, "autocreatepeer")) {
autocreatepeer = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "match_auth_username")) {
+ match_auth_username = ast_true(v->value);
} else if (!strcasecmp(v->name, "srvlookup")) {
srvlookup = ast_true(v->value);
} else if (!strcasecmp(v->name, "pedantic")) {
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