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Merged revisions 274284 via svnmerge from

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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/
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.2
Terry Wilson 16 years ago
parent 0257ac0bf3
commit 29beb317eb
2 changed files with 22 additions and 11 deletions
Show Stats Download Patch File Download Diff File

30
channels/chan_sip.c
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@ -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;

3
configs/sip.conf.sample
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@ -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

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