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Support SIP_CODEC channel variable for early media. (Imported from 1.2, with a small

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change for const char* channel variables)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-03-09 17:41:38 +00:00
parent 0752f8e41e
commit b27fa8bfc7
1 changed files with 26 additions and 15 deletions
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41
channels/chan_sip.c
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@@ -2605,12 +2605,34 @@ static int sip_hangup(struct ast_channel *ast)
return 0;
}
/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
static void try_suggested_sip_codec(struct sip_pvt *p)
{
int fmt;
const char *codec;
codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
if (!codec)
return;
fmt = ast_getformatbyname(codec);
if (fmt) {
ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n", codec);
if (p->jointcapability & fmt) {
p->jointcapability &= fmt;
p->capability &= fmt;
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
return;
}
/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite
* Part of PBX interface */
static int sip_answer(struct ast_channel *ast)
{
int res = 0,fmt;
const char *codec;
int res = 0;
struct sip_pvt *p = ast->tech_pvt;
ast_mutex_lock(&p->lock);
@@ -2618,19 +2640,7 @@ static int sip_answer(struct ast_channel *ast)
#ifdef OSP_SUPPORT
time(&p->ospstart);
#endif
codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
if (codec) {
fmt=ast_getformatbyname(codec);
if (fmt) {
ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
if (p->jointcapability & fmt) {
p->jointcapability &= fmt;
p->capability &= fmt;
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
}
try_suggested_sip_codec(p);
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
@@ -4671,6 +4681,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
}
respprep(&resp, p, msg, req);
if (p->rtp) {
try_suggested_sip_codec(p);
add_sdp(&resp, p);
} else {
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);