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Fix building with -Wdeclaration-after-statement, here too 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72491 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
qwell 2007-06-28 19:35:23 +00:00
parent a1394d0f42
commit 19dd1526bc
1 changed files with 8 additions and 5 deletions
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13
channels/chan_sip.c
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@ -4167,6 +4167,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
int text;
int needvideo = 0;
int needtext = 0;
char buf[BUFSIZ];
{
const char *my_name; /* pick a good name */
@ -4211,7 +4212,6 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
/* Set the native formats for audio and merge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text;
char buf[BUFSIZ];
ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
@ -5184,6 +5184,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int found_rtpmap_codecs[32];
int last_rtpmap_codec=0;
char buf[BUFSIZ];
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1;
@ -5696,7 +5698,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
/* Ok, we're going with this offer */
char buf[BUFSIZ];
ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability));
if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
@ -6737,6 +6738,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
int min_video_packet_size = 0;
int min_text_packet_size = 0;
char codecbuf[BUFSIZ];
char buf[BUFSIZ];
m_video[0] = '\0'; /* Reset the video media string if it's not needed */
m_text[0] = '\0'; /* Reset the video media string if it's not needed */
@ -6756,7 +6760,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
capability = p->jointcapability;
char codecbuf[BUFSIZ];
ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False");
ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
@ -6974,7 +6977,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
/* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
char buf[BUFSIZ];
ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
return AST_SUCCESS;
@ -13898,6 +13900,8 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */
struct ast_channel *targetcall; /* The bridge to the take-over target */
struct ast_channel *test;
/* Check if we're in ring state */
if (replacecall->_state == AST_STATE_RING)
earlyreplace = 1;
@ -14011,7 +14015,6 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
sip_pvt_unlock(p->refer->refer_call);
ast_setstate(c, AST_STATE_DOWN);
struct ast_channel *test;
ast_debug(4, "After transfer:----------------------------\n");
ast_debug(4, " -- C: %s State %s\n", c->name, ast_state2str(c->_state));
if (replacecall)