From: George Joseph <gjoseph@digium.com>
Date: Thu, 7 Jun 2018 13:46:03 +0000 (-0600)
Subject: chan_pjsip:  Register for "BEFORE_MEDIA" responses
X-Git-Tag: 16.0.0-rc1~113^2
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=1725eaf8fba013db84a772dcba37ab2a9837416a;p=thirdparty%2Fasterisk.git

chan_pjsip:  Register for "BEFORE_MEDIA" responses

chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant
it was not updating HANGUPCAUSE for 4XX responses.  If the remote end
sent a "180 Ringing", then a "486 Busy", the hangup cause was left at
"180 Normal Clearing".

* Removed chan_pjsip_incoming_response from the original session
  supplement (which was handling only "AFTER MEDIA") and added it to a
  new session supplement which accepts both "BEFORE_MEDIA" and
  "AFTER_MEDIA".

* Also cleaned up some cleanup code in load module.

ASTERISK-27902

Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a
---

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index b4eb711f79..21941ebfd6 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -144,11 +144,18 @@ static struct ast_sip_session_supplement chan_pjsip_supplement = {
 	.session_begin = chan_pjsip_session_begin,
 	.session_end = chan_pjsip_session_end,
 	.incoming_request = chan_pjsip_incoming_request,
-	.incoming_response = chan_pjsip_incoming_response,
 	/* It is important that this supplement runs after media has been negotiated */
 	.response_priority = AST_SIP_SESSION_AFTER_MEDIA,
 };
 
+/*! \brief SIP session supplement structure just for responses */
+static struct ast_sip_session_supplement chan_pjsip_supplement_response = {
+	.method = "INVITE",
+	.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
+	.incoming_response = chan_pjsip_incoming_response,
+	.response_priority = AST_SIP_SESSION_BEFORE_MEDIA | AST_SIP_SESSION_AFTER_MEDIA,
+};
+
 static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 
 static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
@@ -3109,6 +3116,7 @@ static int load_module(void)
 	}
 
 	ast_sip_session_register_supplement(&chan_pjsip_supplement);
+	ast_sip_session_register_supplement(&chan_pjsip_supplement_response);
 
 	if (!(pjsip_uids_onhold = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_RWLOCK,
 			AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT, 37, uid_hold_hash_fn,
@@ -3123,10 +3131,6 @@ static int load_module(void)
 
 	if (pjsip_channel_cli_register()) {
 		ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
-		ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
-		ast_sip_session_unregister_supplement(&pbx_start_supplement);
-		ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
-		ast_sip_session_unregister_supplement(&call_pickup_supplement);
 		goto end;
 	}
 
@@ -3142,6 +3146,11 @@ static int load_module(void)
 end:
 	ao2_cleanup(pjsip_uids_onhold);
 	pjsip_uids_onhold = NULL;
+	ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
+	ast_sip_session_unregister_supplement(&pbx_start_supplement);
+	ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
+	ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
+	ast_sip_session_unregister_supplement(&call_pickup_supplement);
 	ast_custom_function_unregister(&dtmf_mode_function);
 	ast_custom_function_unregister(&media_offer_function);
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
@@ -3160,6 +3169,7 @@ static int unload_module(void)
 
 	pjsip_channel_cli_unregister();
 
+	ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
 	ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
 	ast_sip_session_unregister_supplement(&pbx_start_supplement);
 	ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);