kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-12 15:45:18 +00:00
Code Issues Packages Projects Releases Wiki Activity

Add channel locking for channel snapshot creation.

Browse Source 
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@403311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2013-12-03 16:33:49 +00:00
parent d904906a1a
commit 4b7ba33125
57 changed files with 384 additions and 31 deletions
Download Patch File Download Diff File Expand all files Collapse all files

13
channels/chan_pjsip.c
View File

@@ -577,13 +577,15 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
return NULL;
}
ast_channel_stage_snapshot(chan);
/* If res_pjsip_session is ever updated to create/destroy ast_sip_session_media
* during a call such as if multiple same-type stream support is introduced,
* these will need to be recaptured as well */
pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
ast_channel_lock(chan);
ast_channel_stage_snapshot(chan);
ast_channel_tech_pvt_set(chan, channel);
if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
@@ -632,9 +634,10 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
ast_channel_zone_set(chan, zone);
}
ast_endpoint_add_channel(session->endpoint->persistent, chan);
ast_channel_stage_snapshot_done(chan);
ast_channel_unlock(chan);
ast_endpoint_add_channel(session->endpoint->persistent, chan);
return chan;
}
@@ -2030,9 +2033,11 @@ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct
switch (status.code) {
case 180:
ast_queue_control(session->channel, AST_CONTROL_RINGING);
ast_channel_lock(session->channel);
if (ast_channel_state(session->channel) != AST_STATE_UP) {
ast_setstate(session->channel, AST_STATE_RINGING);
}
ast_channel_unlock(session->channel);
break;
case 183:
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);