[asterisk-commits] channels/chan sip: Set cause code to 44 on RTP timeout (asterisk[13])

[SVN commits to the Asterisk project]

Matt Jordan has submitted this change and it was merged.

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(-)

Approvals:
  Matt Jordan: Looks good to me, approved; Verified
  Joshua Colp: Looks good to me, but someone else must approve



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

Gerrit-MessageType: merged
Gerrit-Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>