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

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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/1.4@274280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-07-06 22:08:20 +00:00
parent 5724d8f905
commit 79d795c383
2 changed files with 22 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files
channels/chan_sip.c
+19 -10
View File
@@ -812,11 +812,12 @@ struct sip_auth {
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 28) /*!< 28: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 29) /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_RPORT_PRESENT (1 << 30) /*!< 30: Was rport received in the Via header? */
#define SIP_PAGE2_FORWARD_LOOP_DETECTED (1 << 31) /*!< 31: Do call forward when receiving 482 Loop Detected */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
SIP_PAGE2_UDPTL_DESTINATION)
SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_FORWARD_LOOP_DETECTED)
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
@@ -11200,6 +11201,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, const struct m
ast_cli(fd, " Send RPID : %s\n", ast_test_flag(&peer->flags[0], SIP_SENDRPID) ? "Yes" : "No");
ast_cli(fd, " Subscriptions: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
ast_cli(fd, " Overlap dial : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No");
ast_cli(fd, " Forward Loop : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED) ? "Yes" : "No");
/* - is enumerated */
ast_cli(fd, " DTMFmode : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -11503,7 +11505,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
ast_cli(fd, " MOH Interpret: %s\n", default_mohinterpret);
ast_cli(fd, " MOH Suggest: %s\n", default_mohsuggest);
ast_cli(fd, " Voice Mail Extension: %s\n", default_vmexten);
ast_cli(fd, " Forward Detected Loops: %s\n", (ast_test_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED) ? "Yes" : "No"));
if (realtimepeers || realtimeusers) {
ast_cli(fd, "\nRealtime SIP Settings:\n");
@@ -13701,16 +13703,19 @@ 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... */
if (option_debug)
ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
if (p->owner)
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)) {
if (option_debug) {
ast_log(LOG_DEBUG, "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 */
@@ -17570,6 +17575,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} 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;
@@ -18386,6 +18394,7 @@ 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_CAN_REINVITE); /*!< 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;
configs/sip.conf.sample
+3
View File
@@ -176,6 +176,9 @@ 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
;
; If regcontext is specified, Asterisk will dynamically create and destroy a