[Asterisk-code-review] res pjsip sdp rtp: Set hangup cause for RTP timeouts (asterisk[master])

[Joshua Colp]

Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/5318 )

Change subject: res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts
......................................................................


res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts

chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL
(44) when a channel is hung up due to an RTP timeout. So do the same
when it happens with PJSIP for parity.

Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 4 insertions(+), 0 deletions(-)

Approvals:
  Mark Michelson: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, approved



diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index a824757..b66c1ae 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -163,6 +163,10 @@
 	ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of RTP activity in %d seconds\n",
 		ast_channel_name(chan), elapsed);
 
+	ast_channel_lock(chan);
+	ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
+	ast_channel_unlock(chan);
+
 	ast_softhangup(chan, AST_SOFTHANGUP_DEV);
 	ast_channel_unref(chan);
 

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Gerrit-MessageType: merged
Gerrit-Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>