Merged revisions 274284 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
  r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) | 18 lines

  Merged revisions 274280 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

  ........
    r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines

    Add option to not do a call forward on 482 Loop Detected

    Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
    This prevents handling the call failure by just continuing on in the dialplan.
    Since this would be a change in behavior, the new option to disable this
    behavior is forwardloopdetected which defaults to 'yes'.

    Review: https://reviewboard.asterisk.org/r/764/
  ........

................

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274360 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6a83b433456d3b998301c86900ad321f3e25c1fb..0d37cfc0708300e153de547f9d2e3b0aba584e06 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1375,6 +1375,7 @@ struct sip_auth {
 #define SIP_PAGE2_RTAUTOCLEAR          (1 << 2)        /*!< GP: Should we clean memory from peers after expiry? */
 #define SIP_PAGE2_HAVEPEERCONTEXT      (1 << 3)        /*< Are we associated with a configured peer context? */
 /* Space for addition of other realtime flags in the future */
+#define SIP_PAGE2_FORWARD_LOOP_DETECTED (1 << 8)/*!< 31: Do call forward when receiving 482 Loop Detected */
 #define SIP_PAGE2_STATECHANGEQUEUE     (1 << 9)        /*!< D: Unsent state pending change exists */
 
 #define SIP_PAGE2_RPORT_PRESENT         (1 << 10)       /*!< Was rport received in the Via header? */
@@ -1411,7 +1412,7 @@ struct sip_auth {
        SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
        SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
        SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | \
-       SIP_PAGE2_HAVEPEERCONTEXT)
+       SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_FORWARD_LOOP_DETECTED)
 
 /*@}*/ 
 
@@ -15531,6 +15532,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
                ast_cli(fd, "  Send RPID    : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
                ast_cli(fd, "  Subscriptions: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
                ast_cli(fd, "  Overlap dial : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
+               ast_cli(fd, "  Forward Loop : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED)));
                if (peer->outboundproxy)
                        ast_cli(fd, "  Outb. proxy  : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
                                                        peer->outboundproxy->force ? "(forced)" : "");
@@ -16195,7 +16197,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
        ast_cli(a->fd, "  MOH Interpret:          %s\n", default_mohinterpret);
        ast_cli(a->fd, "  MOH Suggest:            %s\n", default_mohsuggest);
        ast_cli(a->fd, "  Voice Mail Extension:   %s\n", default_vmexten);
-
+       ast_cli(a->fd, "  Forward Detected Loops: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED)));
        
        if (realtimepeers || realtimeregs) {
                ast_cli(a->fd, "\nRealtime SIP Settings:\n");
@@ -18618,15 +18620,17 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                                        if (p->owner)
                                                ast_queue_control(p->owner, AST_CONTROL_BUSY);
                                        break;
-                               case 482: /*!
-                                       \note SIP is incapable of performing a hairpin call, which
-                                       is yet another failure of not having a layer 2 (again, YAY
-                                        IETF for thinking ahead).  So we treat this as a call
-                                        forward and hope we end up at the right place... */
-                                       ast_debug(1, "Hairpin detected, setting up call forward for what it's worth\n");
-                                       if (p->owner)
-                                               ast_string_field_build(p->owner, call_forward,
-                                                                      "Local/%s@%s", p->username, p->context);
+                               case 482: /* Loop Detected */
+                                       /*
+                                               \note Asterisk has historically tried to do a call forward when it
+                                               gets a 482, but that behavior isn't necessarily the best course of
+                                               action. Go ahead and do it anyway by default, but allow the option
+                                               to immediately pass to the next line in the dialplan. */
+                                       if (p->owner && ast_test_flag(&p->flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED)) {
+                                                       ast_debug(1, "Hairpin detected, setting up call forward for what it's worth\n");
+                                                       ast_string_field_build(p->owner, call_forward,
+                                                                       "Local/%s@%s", p->username, p->context);
+                                       }
                                        /* Fall through */
                                case 480: /* Temporarily Unavailable */
                                case 404: /* Not Found */
@@ -23390,6 +23394,9 @@ static int handle_t38_options(struct ast_flags *flags, struct ast_flags *mask, s
        } else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
                ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
                ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
+       } else if (!strcasecmp(v->name, "forwardloopdetected")) {
+               ast_set_flag(&mask[1], SIP_PAGE2_FORWARD_LOOP_DETECTED);
+               ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FORWARD_LOOP_DETECTED);
        } else {
                res = 0;
        }
@@ -24614,6 +24621,7 @@ static int reload_config(enum channelreloadreason reason)
        ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833);                       /*!< Default DTMF setting: RFC2833 */
        ast_set_flag(&global_flags[0], SIP_NAT_RFC3581);                        /*!< NAT support if requested by device with rport */
        ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA);                       /*!< Allow re-invites */
+       ast_set_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED); /*!< Set up call forward on 482 Loop Detected */
 
        /* Debugging settings, always default to off */
        dumphistory = FALSE;

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index e47d96f8271468e290ace9c5723ce05224a4e358..0ebca87fc15d56dfdfbeb3b9af85b8d97d70367b 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -301,6 +301,9 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
 ;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                        ; register their phones.
+;forwardloopdetected=no         ; Attempt to forward a call locally if the
+                                ; destination replies with 482 Loop Detected
+                                ; default = yes
 
 ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
 ; in square brackets.  For example, the caller id value 555.5555 becomes 5555555