[asterisk-commits] trunk r33348 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Jun 9 13:09:56 MST 2006
Author: oej
Date: Fri Jun 9 15:09:55 2006
New Revision: 33348
URL: http://svn.digium.com/view/asterisk?rev=33348&view=rev
Log:
Another try at hanging up the transferer channel after the masq
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=33348&r1=33347&r2=33348&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jun 9 15:09:55 2006
@@ -1704,6 +1704,8 @@
if (p->owner) {
ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
ast_queue_hangup(p->owner);
+ } else if (p->refer) {
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
} else {
sip_destroy(p);
}
@@ -2878,12 +2880,13 @@
if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
if (option_debug >3)
- ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup.\n");
+ ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
if (p->autokillid > -1)
sip_cancel_destroy(p);
sip_scheddestroy(p, 32000);
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
ast_clear_flag(&p->flags[0], SIP_NEEDDESTROY);
+ p->owner->tech_pvt = NULL;
p->owner = NULL; /* Owner will be gone after we return, so take it away */
return 0;
}
@@ -3101,7 +3104,7 @@
ast_mutex_lock(&p->lock);
append_history(p, "Masq", "Old channel: %s\n", oldchan->name);
- append_history(p, "Masq (cont)", "...new owner: %s\n", p->owner->name);
+ append_history(p, "Masq (cont)", "...new owner: %s\n", newchan->name);
if (p->owner != oldchan)
ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
else {
@@ -3110,16 +3113,6 @@
}
if (option_debug > 2)
ast_log(LOG_DEBUG, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);
- if (p->refer) {
- if (option_debug > 2) {
- if (oldchan->tech_pvt) {
- struct sip_pvt *old = oldchan->tech_pvt;
- ast_log(LOG_DEBUG, "Releasing connection between %s and pvt %s\n", oldchan->name, old->callid);
- } else
- ast_log(LOG_DEBUG, "Hmmm. No sip_pvt to release for %s\n", oldchan->name);
- }
- oldchan->tech_pvt = NULL; /* Release connection between old channel and it's pvt so we can hang up in peace */
- }
ast_mutex_unlock(&p->lock);
return ret;
@@ -10864,15 +10857,17 @@
/* They got the notify, this is the end */
if (p->owner) {
if (!p->refer) {
- ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
+ ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
ast_queue_hangup(p->owner);
}
} else {
if (p->subscribed == NONE)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
- } else if (sipmethod == SIP_REGISTER)
+ } else if (sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, ignore, seqno);
+ else if (sipmethod == SIP_BYE)
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
break;
case 202: /* Transfer accepted */
if (sipmethod == SIP_REFER)
@@ -11067,7 +11062,7 @@
if (ast_test_flag(req, SIP_PKT_DEBUG))
ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
- if (resp == 200) {
+ if (sipmethod == SIP_INVITE && resp == 200) {
/* Tags in early session is replaced by the tag in 200 OK, which is
the final reply to our INVITE */
char tag[128];
@@ -11087,14 +11082,16 @@
/* They got the notify, this is the end */
if (p->owner) {
ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
- ast_queue_hangup(p->owner);
+ //ast_queue_hangup(p->owner);
} else {
- if (!p->subscribed)
+ if (!p->subscribed && !p->refer)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
- /* Wait for 487, then destroy */
- } else if (sipmethod == SIP_MESSAGE)
+ } else if (sipmethod == SIP_BYE)
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ else if (sipmethod == SIP_MESSAGE)
/* We successfully transmitted a message */
+ /* XXX Why destroy this pvt after message transfer? Bad */
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
break;
case 202: /* Transfer accepted */
@@ -11122,6 +11119,8 @@
if (sipmethod == SIP_INVITE) {
/* Re-invite failed */
handle_response_invite(p, resp, rest, req, seqno);
+ } else if (sipmethod == SIP_BYE) {
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else if (sipdebug) {
ast_log (LOG_DEBUG, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
}
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