[Asterisk-code-review] channels/chan sip: Set cause code to 44 on RTP timeout (asterisk[13])
Matt Jordan
asteriskteam at digium.com
Tue Oct 13 14:25:33 CDT 2015
Matt Jordan has uploaded a new change for review.
https://gerrit.asterisk.org/1433
Change subject: channels/chan_sip: Set cause code to 44 on RTP timeout
......................................................................
channels/chan_sip: Set cause code to 44 on RTP timeout
To quote Olle:
"When issuing a hangup due to RTP timeouts the cause code is not set. I have
selected 44 based on Cisco's implementation..."
ASTERISK-25135 #close
Reported by: Olle Johansson
patches:
rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267)
Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc
---
M channels/chan_sip.c
1 file changed, 2 insertions(+), 1 deletion(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/33/1433/1
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 384e843..051bb2b 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -28806,7 +28806,8 @@
ast_channel_name(dialog->owner), (long) (t - dialog->lastrtprx));
send_session_timeout(dialog->owner, "RTPTimeout");
- /* Issue a softhangup */
+ /* Issue a softhangup - cause 44 (as used by Cisco for RTP timeouts) */
+ ast_channel_hangupcause_set(dialog->owner, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(dialog->owner);
/* forget the timeouts for this call, since a hangup
--
To view, visit https://gerrit.asterisk.org/1433
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Gerrit-MessageType: newchange
Gerrit-Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Matt Jordan <mjordan at digium.com>
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