diff options
| author | Jenkins2 <jenkins2@gerrit.asterisk.org> | 2018-03-12 10:44:46 -0500 |
|---|---|---|
| committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2018-03-12 10:44:46 -0500 |
| commit | 14857195310b1f8f00c89f2b67e3a6c981d39e93 (patch) | |
| tree | 29228ae94a1a62b92478b0e5754209bc7a29e427 /channels | |
| parent | 0d0738aaaf133f524addde9c01735c1aa2312ebb (diff) | |
| parent | c8a521b6c804d66abb367783dba75eecb16889a2 (diff) | |
Merge "Replace direct checks of option_debug with DEBUG_ATLEAST macro."
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_dahdi.c | 4 | ||||
| -rw-r--r-- | channels/chan_sip.c | 14 |
2 files changed, 9 insertions, 9 deletions
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c index de2c15279..0579a5a36 100644 --- a/channels/chan_dahdi.c +++ b/channels/chan_dahdi.c @@ -15058,14 +15058,14 @@ retry: ast_mutex_lock(&p->lock); if (p->owner && !p->restartpending) { if (ast_channel_trylock(p->owner)) { - if (option_debug > 2) + if (DEBUG_ATLEAST(3)) ast_verbose("Avoiding deadlock\n"); /* Avoid deadlock since you're not supposed to lock iflock or pvt before a channel */ ast_mutex_unlock(&p->lock); ast_mutex_unlock(&iflock); goto retry; } - if (option_debug > 2) + if (DEBUG_ATLEAST(3)) ast_verbose("Softhanging up on %s\n", ast_channel_name(p->owner)); ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_EXPLICIT); p->restartpending = 1; diff --git a/channels/chan_sip.c b/channels/chan_sip.c index f0cc2a6bd..cd2929448 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -22479,7 +22479,7 @@ static void sip_dump_history(struct sip_pvt *dialog) return; } - if (!option_debug && !sipdebug) { + if (!sipdebug && !DEBUG_ATLEAST(1)) { if (!errmsg) { ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n"); errmsg = 1; @@ -22487,20 +22487,20 @@ static void sip_dump_history(struct sip_pvt *dialog) return; } - ast_debug(1, "\n---------- SIP HISTORY for '%s' \n", dialog->callid); + ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n", dialog->callid); if (dialog->subscribed) { - ast_debug(1, " * Subscription\n"); + ast_log(LOG_DEBUG, " * Subscription\n"); } else { - ast_debug(1, " * SIP Call\n"); + ast_log(LOG_DEBUG, " * SIP Call\n"); } if (dialog->history) { AST_LIST_TRAVERSE(dialog->history, hist, list) - ast_debug(1, " %-3.3d. %s\n", ++x, hist->event); + ast_log(LOG_DEBUG, " %-3.3d. %s\n", ++x, hist->event); } if (!x) { - ast_debug(1, "Call '%s' has no history\n", dialog->callid); + ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid); } - ast_debug(1, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid); + ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid); } |