[asterisk-commits] oej: trunk r50132 - in /trunk: ./
channels/chan_sip.c
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asterisk-commits at lists.digium.com
Tue Jan 9 05:25:33 MST 2007
Author: oej
Date: Tue Jan 9 06:25:33 2007
New Revision: 50132
URL: http://svn.digium.com/view/asterisk?view=rev&rev=50132
Log:
Based on the following patch, changed for trunk...
Merged revisions 50124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 lines
- handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=50132&r1=50131&r2=50132
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jan 9 06:25:33 2007
@@ -3406,7 +3406,8 @@
return 0;
}
/* If the call is not UP, we need to send CANCEL instead of BYE */
- if (p->invitestate < INV_COMPLETED) {
+ /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
+ if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
needcancel = TRUE;
if (option_debug > 3)
ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
@@ -3756,7 +3757,7 @@
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "486 Busy Here", &p->initreq);
- p->invitestate = INV_TERMINATED;
+ p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
@@ -3766,7 +3767,7 @@
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "503 Service Unavailable", &p->initreq);
- p->invitestate = INV_TERMINATED;
+ p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
@@ -13267,6 +13268,7 @@
/* At this point we only support REPLACES */
transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, required);
ast_log(LOG_WARNING,"Received SIP INVITE with unsupported required extension: %s\n", required);
+ p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
@@ -13281,6 +13283,7 @@
/* If pedantic is on, we need to check the tags. If they're different, this is
in fact a forked call through a SIP proxy somewhere. */
transmit_response(p, "482 Loop Detected", req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
}
@@ -13321,6 +13324,7 @@
transmit_response(p, "500 Server Internal Error", req);
append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ p->invitestate = INV_COMPLETED;
return -1;
}
@@ -13394,6 +13398,7 @@
sip_pvt_unlock(p->refer->refer_call);
ast_channel_unlock(p->refer->refer_call->owner);
}
+ p->invitestate = INV_COMPLETED;
return -1;
}
}
@@ -13446,6 +13451,7 @@
/* Handle authentication if this is our first invite */
res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
if (res == AUTH_CHALLENGE_SENT)
+ p->invitestate = INV_COMPLETED; /* Needs to restart in another INVITE transaction */
return 0;
if (res < 0) { /* Something failed in authentication */
if (res == AUTH_FAKE_AUTH) {
@@ -13455,6 +13461,7 @@
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req);
}
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_free(p, theirtag);
return 0;
@@ -13465,6 +13472,7 @@
if (process_sdp(p, req)) {
/* Unacceptable codecs */
transmit_response_reliable(p, "488 Not acceptable here", req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
if (option_debug)
ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n");
@@ -13495,6 +13503,7 @@
ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ p->invitestate = INV_COMPLETED;
}
return 0;
}
@@ -13513,6 +13522,7 @@
transmit_response_reliable(p, "484 Address Incomplete", req);
else
transmit_response_reliable(p, "404 Not Found", req);
+ p->invitestate = INV_COMPLETED;
update_call_counter(p, DEC_CALL_LIMIT);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
@@ -13713,6 +13723,7 @@
/* No bridged peer with T38 enabled*/
}
}
+ /* Respond to normal re-invite */
if (sendok)
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
@@ -14797,7 +14808,7 @@
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
- p->invitestate = INV_CONFIRMED;
+ p->invitestate = INV_TERMINATED;
p->pendinginvite = 0;
__sip_ack(p, seqno, FLAG_RESPONSE, 0);
if (find_sdp(req)) {
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