[Asterisk-code-review] chan sip: fix Reason-Phrase for 603 Response (asterisk[13])

[Frederic LE FOLL]

Frederic LE FOLL has uploaded this change for review. ( https://gerrit.asterisk.org/10588


Change subject: chan_sip: fix Reason-Phrase for 603 Response
......................................................................

chan_sip: fix Reason-Phrase for 603 Response

While RFC3261 specifies defaut Reason-Pḧrase "Decline" for Response 603
(see https://tools.ietf.org/html/rfc3261#section-21.6.2 "603 Decline"),
chan_sip generates "603 Declined".

Since RFC3261 says "603 Decline", I suggest than chan_sip uses
"603 Decline" instead of "603 Declined".

ASTERISK-28153

Change-Id: I4f94ae98f1cf4302fbad9895cad96eeab8c53269
---
M channels/chan_sip.c
1 file changed, 7 insertions(+), 7 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/88/10588/1

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 97ce93c..5be6953 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7185,7 +7185,7 @@
 	   get an answer to the BYE or INVITE/CANCEL
 	   If we get no answer during retransmit period, drop the call anyway.
 	   (Sorry, mother-in-law, you can't deny a hangup by sending
-	   603 declined to BYE...)
+	   603 decline to BYE...)
 	*/
 	if (p->alreadygone)
 		needdestroy = 1;	/* Set destroy flag at end of this function */
@@ -7224,7 +7224,7 @@
 				if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
 					transmit_response_reliable(p, res, &p->initreq);
 				else
-					transmit_response_reliable(p, "603 Declined", &p->initreq);
+					transmit_response_reliable(p, "603 Decline", &p->initreq);
 				p->invitestate = INV_TERMINATED;
 			}
 		} else {	/* Call is in UP state, send BYE */
@@ -25439,7 +25439,7 @@
 		if the reference was successful, the body:
 			SIP/2.0 503 Service Unavailable
 		if the reference failed, or the body:
-			SIP/2.0 603 Declined
+			SIP/2.0 603 Decline
 
 		if the REFER request was accepted before approval to follow the
 		reference could be obtained and that approval was subsequently denied
@@ -26252,7 +26252,7 @@
 				ast_channel_state(replaces_chan) != AST_STATE_RING &&
 				ast_channel_state(replaces_chan) != AST_STATE_UP) {
 			ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
-			transmit_response_reliable(p, "603 Declined (Replaces)", req);
+			transmit_response_reliable(p, "603 Decline (Replaces)", req);
 			error = 1;
 		}
 
@@ -26996,7 +26996,7 @@
 		/* This is a REFER outside of an existing SIP dialog */
 		/* We can't handle that, so decline it */
 		ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
-		transmit_response(p, "603 Declined (No dialog)", req);
+		transmit_response(p, "603 Decline (No dialog)", req);
 		if (!req->ignore) {
 			append_history(p, "Xfer", "Refer failed. Outside of dialog.");
 			sip_alreadygone(p);
@@ -27008,7 +27008,7 @@
 	/* Check if transfer is allowed from this device */
 	if (p->allowtransfer == TRANSFER_CLOSED ) {
 		/* Transfer not allowed, decline */
-		transmit_response(p, "603 Declined (policy)", req);
+		transmit_response(p, "603 Decline (policy)", req);
 		append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
 		/* Do not destroy SIP session */
 		return 0;
@@ -27042,7 +27042,7 @@
 			}
 			break;
 		case -3:
-			transmit_response(p, "603 Declined (Non sip: uri)", req);
+			transmit_response(p, "603 Decline (Non sip: uri)", req);
 			append_history(p, "Xfer", "Refer failed. Non SIP uri");
 			if (req->debug) {
 				ast_debug(1, "SIP transfer to non-SIP uri denied\n");

-- 
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: newchange
Gerrit-Change-Id: I4f94ae98f1cf4302fbad9895cad96eeab8c53269
Gerrit-Change-Number: 10588
Gerrit-PatchSet: 1
Gerrit-Owner: Frederic LE FOLL <frederic.lefoll at c-s.fr>
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