[svn-commits] twilson: branch 1.4 r274280 - in /branches/1.4: channels/ configs/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 6 17:08:23 CDT 2010
Author: twilson
Date: Tue Jul 6 17:08:20 2010
New Revision: 274280
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=274280
Log:
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/
Modified:
branches/1.4/channels/chan_sip.c
branches/1.4/configs/sip.conf.sample
Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=274280&r1=274279&r2=274280
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jul 6 17:08:20 2010
@@ -812,11 +812,12 @@
#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 @@
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 @@
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 @@
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 @@
} 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 @@
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;
Modified: branches/1.4/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=274280&r1=274279&r2=274280
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Tue Jul 6 17:08:20 2010
@@ -176,6 +176,9 @@
;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
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