[Asterisk-code-review] Fix problem where a hung channel could occur on a failed bli... (asterisk[certified/13.1])

[Kevin Harwell]

Kevin Harwell has uploaded a new change for review.

  https://gerrit.asterisk.org/976

Change subject: Fix problem where a hung channel could occur on a failed blind transfer.
......................................................................

Fix problem where a hung channel could occur on a failed blind transfer.

Different clients react differently to being told that a blind transfer
has failed. Some will simply send a BYE and be done with it. Others will
attempt to reinvite themselves back onto the call.

In the latter case, we were creating a new channel and then leaving it to
sit forever doing nothing. With this code change, that new channel will
not be created and the dialog with the transferring channel will be cleaned
up properly.

ASTERISK-24624 #close
Reported by Zane Conkle

Review: https://reviewboard.asterisk.org/r/4339

Change-Id: I76e440e08e603c1eea40a14951e7b171c0472a55
---
M channels/chan_pjsip.c
M res/res_pjsip_session.c
2 files changed, 17 insertions(+), 1 deletion(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/76/976/1

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 7617ae0..ac73375 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -2025,6 +2025,21 @@
 		return 0;
 	}
 
+	if (session->inv_session->state >= PJSIP_INV_STATE_CONFIRMED) {
+		/* Weird case. We've received a reinvite but we don't have a channel. The most
+		 * typical case for this happening is that a blind transfer fails, and so the
+		 * transferer attempts to reinvite himself back into the call. We already got
+		 * rid of that channel, and the other side of the call is unrecoverable.
+		 *
+		 * We treat this as a failure, so our best bet is to just hang this call
+		 * up and not create a new channel. Clearing defer_terminate here ensures that
+		 * calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
+		 */
+		session->defer_terminate = 0;
+		ast_sip_session_terminate(session, 400);
+		return -1;
+	}
+
 	datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
 	if (!datastore) {
 		return -1;
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 9bcd311..bedb606 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -808,7 +808,8 @@
 
 	if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD ||
 		!(dlg = pjsip_ua_find_dialog(&rdata->msg_info.cid->id, &rdata->msg_info.to->tag, &rdata->msg_info.from->tag, PJ_FALSE)) ||
-		!(session = ast_sip_dialog_get_session(dlg))) {
+		!(session = ast_sip_dialog_get_session(dlg)) ||
+		!session->channel) {
 		return PJ_FALSE;
 	}
 

-- 
To view, visit https://gerrit.asterisk.org/976
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newchange
Gerrit-Change-Id: I76e440e08e603c1eea40a14951e7b171c0472a55
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>