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The following patch with some changes for trunk...

Browse Source 
Merged revisions 126516 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines

Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.

(closes issue #12951)

Reported by: tsearle
Patches: 
      busy_retransmit.patch uploaded by tsearle (license 373)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.1
Olle Johansson 17 years ago
parent dc5c54eaf2
commit 33a54ee23b
1 changed files with 21 additions and 42 deletions
Show Stats Download Patch File Download Diff File

63
channels/chan_sip.c
Unescape View File

@ -5441,7 +5441,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
break;
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "486 Busy Here", &p->initreq);
transmit_response_reliable(p, "486 Busy Here", &p->initreq);
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@ -5451,7 +5451,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
break;
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "503 Service Unavailable", &p->initreq);
transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@ -8026,7 +8026,7 @@ static int transmit_response_with_minse(struct sip_pvt *p, const char *msg, cons
*/
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req)
{
return __transmit_response(p, msg, req, XMIT_CRITICAL);
return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}
/*! \brief Append date to SIP message */
@ -17521,7 +17521,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
being able to call yourself */
/* 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);
transmit_response_reliable(p, "482 Loop Detected", req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
@ -17529,7 +17529,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (!req->ignore && p->pendinginvite) {
/* We already have a pending invite. Sorry. You are on hold. */
transmit_response(p, "491 Request Pending", req);
transmit_response_reliable(p, "491 Request Pending", req);
ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
/* Don't destroy dialog here */
return 0;
@ -17546,7 +17546,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (p->owner) {
ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
transmit_response(p, "400 Bad request", req); /* The best way to not not accept the transfer */
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
/* Do not destroy existing call */
return -1;
}
@ -17558,7 +17558,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_uri_decode(replace_id);
if (!p->refer && !sip_refer_allocate(p)) {
transmit_response(p, "500 Server Internal Error", req);
transmit_response_reliable(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;
@ -17596,7 +17596,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
*/
if ((p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
transmit_response(p, "481 Call Leg Does Not Exist (Replaces)", req);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
}
@ -17609,7 +17609,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (p->refer->refer_call == p) {
ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
transmit_response(p, "400 Bad request", req); /* The best way to not not accept the transfer */
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
error = 1;
}
@ -17617,13 +17617,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
/* Oops, someting wrong anyway, no owner, no call */
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
/* Check for better return code */
transmit_response(p, "481 Call Leg Does Not Exist (Replace)", req);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
error = 1;
}
if (!error && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP ) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
transmit_response(p, "603 Declined (Replaces)", req);
transmit_response_reliable(p, "603 Declined (Replaces)", req);
error = 1;
}
@ -17666,7 +17666,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
/* Handle SDP here if we already have an owner */
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_INITIATE)) {
transmit_response(p, "488 Not acceptable here", req);
transmit_response_reliable(p, "488 Not acceptable here", req);
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
@ -17812,7 +17812,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (!ast_strlen_zero(p_uac_se_hdr)) {
rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref);
if (rtn != 0) {
transmit_response(p, "400 Session-Expires Invalid Syntax", req);
transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@ -17826,7 +17826,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (!ast_strlen_zero(p_uac_min_se)) {
rtn = parse_minse(p_uac_min_se, &uac_min_se);
if (rtn != 0) {
transmit_response(p, "400 Min-SE Invalid Syntax", req);
transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@ -17954,18 +17954,12 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
p->invitestate = INV_COMPLETED;
if (req->ignore)
transmit_response(p, "503 Unavailable", req);
else
transmit_response_reliable(p, "503 Unavailable", req);
transmit_response_reliable(p, "503 Unavailable", req);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
if (req->ignore)
transmit_response(p, "480 Temporarily Unavailable", req);
else
transmit_response_reliable(p, "480 Temporarily Unavailable", req);
transmit_response_reliable(p, "480 Temporarily Unavailable", req);
break;
case AST_PBX_SUCCESS:
/* nothing to do */
@ -17986,10 +17980,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
*nounlock = 1;
if (ast_pickup_call(c)) {
ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
if (req->ignore)
transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p, "503 Unavailable", req);
transmit_response_reliable(p, "503 Unavailable", req);
sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
sip_pvt_unlock(p);
@ -18036,10 +18027,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
sip_pvt_lock(bridgepvt);
change_t38_state(bridgepvt, T38_DISABLED);
sip_pvt_unlock(bridgepvt);
if (req->ignore)
transmit_response(p, "488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
transmit_response_reliable(p, "488 Not acceptable here", req);
}
} else {
@ -18049,10 +18037,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
}
} else {
/* Other side is not a SIP channel */
if (req->ignore)
transmit_response(p, "488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
transmit_response_reliable(p, "488 Not acceptable here", req);
change_t38_state(p, T38_DISABLED);
if (!p->lastinvite) /* Only destroy if this is *not* a re-invite */
@ -18077,10 +18062,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (bridgepvt->t38.state == T38_ENABLED) {
ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
if (req->ignore)
transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
else
transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
sendok = FALSE;
}
/* No bridged peer with T38 enabled*/
@ -18109,10 +18091,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
msg = "503 Unavailable";
}
if (req->ignore)
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
transmit_response_reliable(p, msg, req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}

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