diff options
| author | Kinsey Moore <kmoore@digium.com> | 2012-01-20 21:26:50 +0000 |
|---|---|---|
| committer | Kinsey Moore <kmoore@digium.com> | 2012-01-20 21:26:50 +0000 |
| commit | c6fd4f5d7401633649bbf2b45f57f3ddc6ae18f1 (patch) | |
| tree | 974e01b8dd5f1c0a10396b9df1cdb815738be667 | |
| parent | 778fa4abafbb19eeb7690c51d886908c0040cf9c (diff) | |
SIP session timeout AMI event
Add an AMI event in the Call category that is issued when a call is terminated
due to either RTP stream inactivity or SIP session timer expiration.
Event description:
Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id
`source` can be either RTPTimeout or SIPSessionTimer
(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | UPGRADE.txt | 3 | ||||
| -rw-r--r-- | channels/chan_sip.c | 4 |
2 files changed, 7 insertions, 0 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index e5a81a98d..83c2b2b6c 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -52,6 +52,9 @@ SIP === - A new option "tonezone" for setting default tonezone for the channel driver or individual devices + - A new manager event, "SessionTimeout" has been added and is triggered when + a call is terminated due to RTP stream inactivity or SIP session timer + expiration. users.conf: - A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 1ea54538b..e80d2d913 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -26395,6 +26395,8 @@ static int check_rtp_timeout(struct sip_pvt *dialog, time_t t) } ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n", ast_channel_name(dialog->owner), (long) (t - dialog->lastrtprx)); + manager_event(EVENT_FLAG_CALL, "SessionTimeout", "Source: RTPTimeout\r\n" + "Channel: %s\r\nUniqueid: %s\r\n", ast_channel_name(dialog->owner), dialog->owner->uniqueid); /* Issue a softhangup */ ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV); ast_channel_unlock(dialog->owner); @@ -26647,6 +26649,8 @@ static int proc_session_timer(const void *vp) sip_pvt_lock(p); } + manager_event(EVENT_FLAG_CALL, "SessionTimeout", "Source: SIPSessionTimer\r\n" + "Channel: %s\r\nUniqueid: %s\r\n", ast_channel_name(p->owner), p->owner->uniqueid); ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV); ast_channel_unlock(p->owner); sip_pvt_unlock(p); |