[svn-commits] russell: branch 1.8 r283527 - /branches/1.8/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Aug 25 09:55:07 CDT 2010
Author: russell
Date: Wed Aug 25 09:55:00 2010
New Revision: 283527
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283527
Log:
Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
Modified:
branches/1.8/channels/chan_sip.c
Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=283527&r1=283526&r2=283527
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Wed Aug 25 09:55:00 2010
@@ -5270,8 +5270,7 @@
if (p->owner) {
if (lockowner)
ast_channel_lock(p->owner);
- if (option_debug)
- ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
+ ast_debug(1, "Detaching from %s\n", p->owner->name);
p->owner->tech_pvt = NULL;
/* Make sure that the channel knows its backend is going away */
p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
@@ -6431,7 +6430,7 @@
* just says they are waiting to get AOC-E before completely tearing
* the call down. Since SIP does not support this at the moment go
* ahead and terminate the call here to avoid an unnecessary timeout. */
- ast_log(LOG_DEBUG, "AOC-E termination request received on %s. This is not yet supported on sip. Continue with hangup \n", p->owner->name);
+ ast_debug(1, "AOC-E termination request received on %s. This is not yet supported on sip. Continue with hangup \n", p->owner->name);
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
}
break;
@@ -6897,8 +6896,7 @@
f = ast_dsp_process(p->owner, p->dsp, f);
if (f && f->frametype == AST_FRAME_DTMF) {
if (f->subclass.integer == 'f') {
- if (option_debug)
- ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
+ ast_debug(1, "Fax CNG detected on %s\n", ast->name);
*faxdetect = 1;
/* If we only needed this DSP for fax detection purposes we can just drop it now */
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
@@ -8121,8 +8119,7 @@
break;
}
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED.");
+ ast_debug(3, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED.");
}
@@ -8306,11 +8303,10 @@
break;
}
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Processing media-level (%s) SDP %c=%s... %s\n",
- (audio == TRUE)? "audio" : (video == TRUE)? "video" : "image",
- type, value,
- (processed == TRUE)? "OK." : "UNSUPPORTED.");
+ ast_debug(3, "Processing media-level (%s) SDP %c=%s... %s\n",
+ (audio == TRUE)? "audio" : (video == TRUE)? "video" : "image",
+ type, value,
+ (processed == TRUE)? "OK." : "UNSUPPORTED.");
}
}
@@ -8486,10 +8482,8 @@
if (udptlportno > 0) {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
ast_rtp_instance_get_remote_address(p->rtp, isa);
- if (!ast_sockaddr_isnull(isa)) {
- if (debug) {
- ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_sockaddr_stringify(isa));
- }
+ if (!ast_sockaddr_isnull(isa) && debug) {
+ ast_debug(1, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_sockaddr_stringify(isa));
}
}
ast_sockaddr_set_port(isa, udptlportno);
@@ -8765,8 +8759,7 @@
struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
if (!format.asterisk_format || !format.code) /* non-codec or not found */
continue;
- if (option_debug)
- ast_log(LOG_DEBUG, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
+ ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
ast_codec_pref_setsize(pref, format.code, framing);
}
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
@@ -12032,7 +12025,7 @@
send_response(p, &resp, XMIT_UNRELIABLE, 0);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
} else {
- ast_log(LOG_DEBUG, "Unable able to send update to '%s' in state '%s'\n", p->owner->name, ast_state2str(p->owner->_state));
+ ast_debug(1, "Unable able to send update to '%s' in state '%s'\n", p->owner->name, ast_state2str(p->owner->_state));
}
} else {
ast_set_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
@@ -23353,8 +23346,7 @@
return 0;
}
if (p->ocseq && (p->ocseq < seqno)) {
- if (option_debug)
- ast_log(LOG_DEBUG, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq);
+ ast_debug(1, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq);
return -1;
} else {
char causevar[256], causeval[256];
@@ -23546,9 +23538,7 @@
while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
if (handle_incoming(p, req, &p->recv, recount, nounlock) == -1) {
/* Request failed */
- if (option_debug) {
- ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
- }
+ ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
ast_free(req);
}
@@ -27550,7 +27540,7 @@
if (strncasecmp(ast_var_name(newvariable), "SIPADDHEADER", strlen("SIPADDHEADER")) == 0) {
if (removeall || (!strncasecmp(ast_var_value(newvariable),inbuf,strlen(inbuf)))) {
if (sipdebug)
- ast_log(LOG_DEBUG,"removing SIP Header \"%s\" as %s\n",
+ ast_debug(1,"removing SIP Header \"%s\" as %s\n",
ast_var_value(newvariable),
ast_var_name(newvariable));
AST_LIST_REMOVE_CURRENT(entries);
@@ -27826,7 +27816,7 @@
return 1;
}
if (addrs_cnt > 1) {
- ast_log(LOG_DEBUG, "Multiple addresses, using the first one only\n");
+ ast_debug(1, "Multiple addresses, using the first one only\n");
}
ast_sockaddr_copy(addr, &addrs[0]);
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