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Sun Feb 23 07:00:00 CET 2003

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matteo Brancaleoni
2003-02-23 06:00:11 +00:00
parent 877a281b74
commit 17769619c2
20 changed files with 1088 additions and 238 deletions
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29
channels/chan_sip.c
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@@ -698,13 +698,38 @@ static int sip_hangup(struct ast_channel *ast)
static int sip_answer(struct ast_channel *ast)
{
int res = 0;
int res = 0,fmt,capability;
char *codec;
struct sip_pvt *p = ast->pvt->pvt;
struct sip_codec_pref *oldpref=NULL;
if (ast->_state != AST_STATE_UP) {
codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
if (codec) {
ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
fmt=ast_getformatbyname(codec);
if (fmt) {
oldpref=prefs;
prefs=NULL;
sip_pref_append(fmt);
capability=p->capability;
p->capability=fmt;
} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized codec: %s\n",codec);
}
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "sip_answer(%s)\n", ast->name);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq);
sip_prefs_free();
if (oldpref) {
prefs=oldpref;
p->capability=capability;
}
}
return res;
}
@@ -1676,7 +1701,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
if ((codec = ast2rtp(cur->codec)) > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strcat(m, costr);
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast2rtpn(x));
snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast2rtpn(rtp2ast(codec)));
strcat(a, costr);
}
}