[Asterisk-code-review] channels/chan sip: Set cause code to 44 on RTP timeout (asterisk[master])

Matt Jordan asteriskteam at digium.com
Tue Oct 13 14:27:57 CDT 2015


Matt Jordan has uploaded a new change for review.

  https://gerrit.asterisk.org/1434

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/34/1434/1

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 349042c..3fdc3ca 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -28811,7 +28811,8 @@
 					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/1434
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newchange
Gerrit-Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Matt Jordan <mjordan at digium.com>



More information about the asterisk-code-review mailing list