[Asterisk-code-review] Fix closing rtp ports after call finished in chan unistim. (asterisk[14])
Igor Goncharovsky
asteriskteam at digium.com
Fri Nov 11 22:01:48 CST 2016
Igor Goncharovsky has uploaded a new change for review. ( https://gerrit.asterisk.org/4401 )
Change subject: Fix closing rtp ports after call finished in chan_unistim.
......................................................................
Fix closing rtp ports after call finished in chan_unistim.
Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.
Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc
---
M channels/chan_unistim.c
1 file changed, 6 insertions(+), 5 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/01/4401/1
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 5b64f96..23e7006 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -4129,7 +4129,7 @@
send_date_time2(pte);
send_idle_clock(pte);
if (strlen(pte->device->maintext0)) {
- send_text(TEXT_LINE0, TEXT_NORMAL, pte, pte->device->maintext0);
+ send_text(TEXT_LINE0, TEXT_NORMAL, pte, ustmtext(pte->device->maintext0, pte));
}
} else {
if (pte->device->missed_call == 1) {
@@ -4148,11 +4148,11 @@
strcat(tmpbuf, ast_inet_ntoa(pte->sin.sin_addr));
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmpbuf);
} else {
- send_text(TEXT_LINE2, TEXT_NORMAL, pte, pte->device->maintext2);
+ send_text(TEXT_LINE2, TEXT_NORMAL, pte, ustmtext(pte->device->maintext2, pte));
}
}
- send_texttitle(pte, pte->device->titledefault);
+ send_texttitle(pte, ustmtext(pte->device->titledefault, pte));
change_favorite_icon(pte, FAV_LINE_ICON);
}
@@ -4407,7 +4407,7 @@
strcat(tmp, pte->macaddr);
send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmp);
send_text_status(pte, "");
- send_texttitle(pte, "UNISTIM for*");
+ send_texttitle(pte, ustmtext("UNISTIM for*", pte));
return;
}
}
@@ -4897,14 +4897,15 @@
ast_channel_tech_pvt_set(ast, NULL);
unistim_set_owner(sub, NULL);
sub->alreadygone = 0;
- ast_mutex_unlock(&sub->lock);
if (sub->rtp) {
if (unistimdebug) {
ast_verb(0, "Destroying RTP session\n");
}
+ ast_rtp_instance_stop(sub->rtp);
ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
+ ast_mutex_unlock(&sub->lock);
return 0;
}
--
To view, visit https://gerrit.asterisk.org/4401
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Igor Goncharovsky <igor.goncharovsky at gmail.com>
More information about the asterisk-code-review
mailing list