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Merged revisions 283527 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines
  
  Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283528 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
russell 2010-08-25 14:55:47 +00:00
parent 65c7bb2346
commit 0dc2051939
1 changed files with 16 additions and 26 deletions
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42
channels/chan_sip.c
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@ -5270,8 +5270,7 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
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 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
* 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 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
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) {
@ -8124,8 +8122,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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.");
}
@ -8309,11 +8306,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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.");
}
}
@ -8489,10 +8485,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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);
@ -8768,8 +8762,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
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);
@ -12035,7 +12028,7 @@ static void update_connectedline(struct sip_pvt *p, const void *data, size_t dat
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);
@ -23385,8 +23378,7 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
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];
@ -23578,9 +23570,7 @@ static void process_request_queue(struct sip_pvt *p, int *recount, int *nounlock
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);
}
@ -27582,7 +27572,7 @@ static int sip_removeheader(struct ast_channel *chan, const char *data)
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);
@ -27858,7 +27848,7 @@ static int ast_sockaddr_resolve_first_af(struct ast_sockaddr *addr,
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]);