[svn-commits] twilson: branch 1.6.2 r274360 - in /branches/1.6.2: channels/ configs/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 6 18:06:36 CDT 2010
Author: twilson
Date: Tue Jul 6 18:06:32 2010
New Revision: 274360
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=274360
Log:
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/
........
................
Modified:
branches/1.6.2/channels/chan_sip.c
branches/1.6.2/configs/sip.conf.sample
Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=274360&r1=274359&r2=274360
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Tue Jul 6 18:06:32 2010
@@ -1375,6 +1375,7 @@
#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 @@
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 @@
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 @@
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 @@
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 @@
} 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 @@
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;
Modified: branches/1.6.2/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/configs/sip.conf.sample?view=diff&rev=274360&r1=274359&r2=274360
==============================================================================
--- branches/1.6.2/configs/sip.conf.sample (original)
+++ branches/1.6.2/configs/sip.conf.sample Tue Jul 6 18:06:32 2010
@@ -301,6 +301,9 @@
;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
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
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