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authorzuul <zuul@gerrit.asterisk.org>2017-03-24 19:21:51 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-03-24 19:21:51 -0500
commit39fff2ebf9356eaceac6aa0c783a8616cc3b9491 (patch)
treee72ee654a27c22bde73af1f20ee590af07ca8a8a
parentddbff36c1259307a27ac2eb72424457f314fc228 (diff)
parentd9d2beba1c0d9aa93382dc4ec8b54627b2a301fa (diff)
Merge "res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts" into 13
-rw-r--r--res/res_pjsip_sdp_rtp.c4
1 files changed, 4 insertions, 0 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index d44171cf8..52ae65cc7 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -163,6 +163,10 @@ static int rtp_check_timeout(const void *data)
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);