Implement a setting for denying/allowing transfer requests. At this stage,

we only have open/closed. Well, at least you can deny transfers from unknown
callers or at least incoming calls.
(Part of the SIPtransfer branch)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9241317..480d952 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -203,6 +203,15 @@ static int usecnt = 0;
 #define RTP    1
 #define NO_RTP 0
 
+/*! \brief Authorization scheme for call transfers 
+\note Not a bitfield flag, since there are plans for other modes,
+       like "only allow transfers for authenticated devices" */
+enum transfermodes {
+       TRANSFER_OPENFORALL,            /*!< Allow all SIP transfers */
+       TRANSFER_CLOSED,                /*!< Allow no SIP transfers */
+};
+
+
 /* Do _NOT_ make any changes to this enum, or the array following it;
    if you think you are doing the right thing, you are probably
    not doing the right thing. If you think there are changes
@@ -450,6 +459,7 @@ static char global_useragent[AST_MAX_EXTENSION];    /*!< Useragent for the SIP chan
 static int allow_external_domains;     /*!< Accept calls to external SIP domains? */
 static int global_callevents;          /*!< Whether we send manager events or not */
 static int global_t1min;               /*!< T1 roundtrip time minimum */
+enum transfermodes global_allowtransfer;       /*! SIP Refer restriction scheme */
 
 /*! \brief Codecs that we support by default: */
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
@@ -756,6 +766,7 @@ static struct sip_pvt {
        int rtptimeout;                         /*!< RTP timeout time */
        int rtpholdtimeout;                     /*!< RTP timeout when on hold */
        int rtpkeepalive;                       /*!< Send RTP packets for keepalive */
+       enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
        enum subscriptiontype subscribed;       /*!< Is this dialog a subscription?  */
        int stateid;
        int laststate;                          /*!< Last known extension state */
@@ -817,6 +828,7 @@ struct sip_user {
        int capability;                 /*!< Codec capability */
        int inUse;                      /*!< Number of calls in use */
        int call_limit;                 /*!< Limit of concurrent calls */
+       enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
        struct ast_ha *ha;              /*!< ACL setting */
        struct ast_variable *chanvars;  /*!< Variables to set for channel created by user */
        int maxcallbitrate;             /*!< Maximum Bitrate for a video call */
@@ -845,6 +857,7 @@ struct sip_peer {
        int callingpres;                /*!< Calling id presentation */
        int inUse;                      /*!< Number of calls in use */
        int call_limit;                 /*!< Limit of concurrent calls */
+       enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
        char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
        char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
        char language[MAX_LANGUAGE];    /*!<  Default language for prompts */
@@ -2001,6 +2014,7 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
        r->maxtime = peer->maxms;
        r->callgroup = peer->callgroup;
        r->pickupgroup = peer->pickupgroup;
+       r->allowtransfer = peer->allowtransfer;
        /* Set timer T1 to RTT for this peer (if known by qualify=) */
        /* Minimum is settable or default to 100 ms */
        if (peer->maxms && peer->lastms)
@@ -3293,6 +3307,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
        /* Assign default music on hold class */
        ast_string_field_set(p, musicclass, default_musicclass);
        p->capability = global_capability;
+       p->allowtransfer = global_allowtransfer;
        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;
@@ -7210,6 +7225,7 @@ static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipme
                        ast_string_field_set(p, accountcode, user->accountcode);
                        ast_string_field_set(p, language, user->language);
                        ast_string_field_set(p, musicclass, user->musicclass);
+                       p->allowtransfer = user->allowtransfer;
                        p->amaflags = user->amaflags;
                        p->callgroup = user->callgroup;
                        p->pickupgroup = user->pickupgroup;
@@ -7495,6 +7511,16 @@ static int sip_show_inuse(int fd, int argc, char *argv[]) {
 #undef FORMAT2
 }
 
+/*! \brief Convert transfer mode to text string */
+static char *transfermode2str(enum transfermodes mode)
+{
+       if (mode == TRANSFER_OPENFORALL)
+               return "open";
+       else if (mode == TRANSFER_CLOSED)
+               return "closed";
+       return "strict";
+}
+
 /*! \brief  Convert NAT setting to text string */
 static char *nat2str(int nat)
 {
@@ -8100,6 +8126,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message
                if (!ast_strlen_zero(peer->accountcode))
                        ast_cli(fd, "  Accountcode  : %s\n", peer->accountcode);
                ast_cli(fd, "  AMA flags    : %s\n", ast_cdr_flags2str(peer->amaflags));
+               ast_cli(fd, "  Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
                ast_cli(fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(peer->callingpres));
                if (!ast_strlen_zero(peer->fromuser))
                        ast_cli(fd, "  FromUser     : %s\n", peer->fromuser);
@@ -8189,6 +8216,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, struct message
                astman_append(s, "Pickupgroup: ");
                print_group(fd, peer->pickupgroup, 1);
                astman_append(s, "VoiceMailbox: %s\r\n", peer->mailbox);
+               astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
                astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
                astman_append(s, "Call limit: %d\r\n", peer->call_limit);
                astman_append(s, "MaxCallBR: %dkbps\r\n", peer->maxcallbitrate);
@@ -8275,6 +8303,7 @@ static int sip_show_user(int fd, int argc, char *argv[])
                if (!ast_strlen_zero(user->accountcode))
                        ast_cli(fd, "  Accountcode  : %s\n", user->accountcode);
                ast_cli(fd, "  AMA flags    : %s\n", ast_cdr_flags2str(user->amaflags));
+               ast_cli(fd, "  Transfer mode: %s\n", transfermode2str(user->allowtransfer));
                ast_cli(fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(user->callingpres));
                ast_cli(fd, "  Call limit   : %d\n", user->call_limit);
                ast_cli(fd, "  Callgroup    : ");
@@ -8395,6 +8424,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
        ast_cli(fd, "  Outbound reg. timeout:  %d secs\n", global_reg_timeout);
        ast_cli(fd, "  Outbound reg. attempts: %d\n", global_regattempts_max);
        ast_cli(fd, "  Notify ringing state:   %s\n", global_notifyringing ? "Yes" : "No");
+       ast_cli(fd, "  SIP Transfer mode:      %s\n", transfermode2str(global_allowtransfer));
        ast_cli(fd, "  Max Call Bitrate:       %dkbps\r\n", default_maxcallbitrate);
        ast_cli(fd, "\nDefault Settings:\n");
        ast_cli(fd, "-----------------\n");
@@ -8672,6 +8702,7 @@ static int sip_show_channel(int fd, int argc, char *argv[])
                        ast_cli(fd, "  Format                  %s\n", ast_getformatname(cur->owner ? cur->owner->nativeformats : 0) );
                        ast_cli(fd, "  Theoretical Address:    %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), ntohs(cur->sa.sin_port));
                        ast_cli(fd, "  Received Address:       %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->recv.sin_addr), ntohs(cur->recv.sin_port));
+                       ast_cli(fd, "  SIP Transfer mode:      %s\n", transfermode2str(cur->allowtransfer));
                        ast_cli(fd, "  NAT Support:            %s\n", nat2str(ast_test_flag(&cur->flags[0], SIP_NAT)));
                        ast_cli(fd, "  Audio IP:               %s %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" );
                        ast_cli(fd, "  Our Tag:                %s\n", cur->tag);
@@ -10933,6 +10964,30 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
 
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "SIP call transfer received for call %s (REFER)!\n", p->callid);
+
+       /* Check if transfer is allowed from this device */
+       if (p->allowtransfer == TRANSFER_CLOSED ) {
+               /* Transfer not allowed, decline */
+               transmit_response(p, "603 Declined (policy)", req);
+               append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
+               /* Do not destroy SIP session */
+               return 0;
+       }
+
+       if (!p->owner) {
+               /* This is a REFER outside of an existing SIP dialog */
+               /* We can't handle that, so decline it */
+               if (option_debug > 2)
+                       ast_log(LOG_DEBUG, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
+               transmit_response(p, "603 Declined (No dialog)", req);
+               if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
+                       append_history(p, "Xfer", "Refer failed. Outside of dialog.");
+                       ast_set_flag(&p->flags[0], SIP_ALREADYGONE);    
+                       ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
+               }
+               return 0;
+       }       
+
        if (ast_strlen_zero(p->context))
                ast_string_field_set(p, context, default_context);
        res = get_refer_info(p, req);
@@ -10945,7 +11000,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                transmit_response_with_allow(p, "404 Not Found", req, 1);
        else {
                int nobye = 0;
-               if (!ignore) {
+               if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
                        if (p->refer_call) {
                                ast_log(LOG_DEBUG,"202 Accepted (supervised)\n");
                                attempt_transfer(p, p->refer_call);
@@ -12377,6 +12432,7 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
        ast_copy_flags(&user->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
        ast_copy_flags(&user->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
        user->capability = global_capability;
+       user->allowtransfer = global_allowtransfer;
        user->prefs = default_prefs;
        /* set default context */
        strcpy(user->context, default_context);
@@ -12402,6 +12458,8 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
                } else if (!strcasecmp(v->name, "permit") ||
                                   !strcasecmp(v->name, "deny")) {
                        user->ha = ast_append_ha(v->name, v->value, user->ha);
+               } else if (!strcasecmp(v->name, "allowtransfer")) {
+                       user->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
                } else if (!strcasecmp(v->name, "secret")) {
                        ast_copy_string(user->secret, v->value, sizeof(user->secret)); 
                } else if (!strcasecmp(v->name, "md5secret")) {
@@ -12475,6 +12533,7 @@ static void set_peer_defaults(struct sip_peer *peer)
        peer->rtptimeout = global_rtptimeout;
        peer->rtpholdtimeout = global_rtpholdtimeout;
        peer->rtpkeepalive = global_rtpkeepalive;
+       peer->allowtransfer = global_allowtransfer;
        strcpy(peer->vmexten, default_vmexten);
        peer->secret[0] = '\0';
        peer->md5secret[0] = '\0';
@@ -12677,6 +12736,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
                        ast_copy_string(peer->vmexten, v->value, sizeof(peer->vmexten));
                } else if (!strcasecmp(v->name, "callgroup")) {
                        peer->callgroup = ast_get_group(v->value);
+               } else if (!strcasecmp(v->name, "allowtransfer")) {
+                       peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
                } else if (!strcasecmp(v->name, "pickupgroup")) {
                        peer->pickupgroup = ast_get_group(v->value);
                } else if (!strcasecmp(v->name, "allow")) {
@@ -12821,6 +12882,7 @@ static int reload_config(enum channelreloadreason reason)
        global_rtptimeout = 0;
        global_rtpholdtimeout = 0;
        global_rtpkeepalive = 0;
+       global_allowtransfer = TRANSFER_OPENFORALL;     /* Merrily accept all transfers by default */
        global_rtautoclear = 120;
        ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE);       /* Default for peers, users: TRUE */
        ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP);         /* Default for peers, users: TRUE */
@@ -12862,7 +12924,10 @@ static int reload_config(enum channelreloadreason reason)
                        ast_copy_string(global_realm, v->value, sizeof(global_realm));
                } else if (!strcasecmp(v->name, "useragent")) {
                        ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
-                       ast_log(LOG_DEBUG, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
+                       if (option_debug)
+                               ast_log(LOG_DEBUG, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
+               } else if (!strcasecmp(v->name, "allowtransfer")) {
+                       global_allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
                } else if (!strcasecmp(v->name, "rtcachefriends")) {
                        ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);  
                } else if (!strcasecmp(v->name, "rtupdate")) {