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automerge commit

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@12538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Automerge script
2006-03-12 16:08:23 +00:00
parent eb6fc20dc8
commit ba70bebc38
1 changed files with 39 additions and 22 deletions
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61
channels/chan_sip.c
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@@ -2485,13 +2485,15 @@ static int sip_hangup(struct ast_channel *ast)
if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
if (needcancel) { /* Outgoing call, not up */
if (ast_test_flag(p, SIP_OUTGOING)) {
/* stop retransmitting an INVITE that has not received a response */
__sip_pretend_ack(p);
/* Send a new request: CANCEL */
transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
ast_clear_flag(&locflags, SIP_NEEDDESTROY);
sip_scheddestroy(p, 15000);
/* stop retransmitting an INVITE that has not received a response */
__sip_pretend_ack(p);
sip_scheddestroy(p, 32000);
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
only if the channel is not auto-congested */
@@ -2521,12 +2523,34 @@ 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;
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;
char *codec;
int res = 0;
struct sip_pvt *p = ast->tech_pvt;
ast_mutex_lock(&p->lock);
@@ -2534,19 +2558,7 @@ 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)
@@ -4580,6 +4592,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
respprep(&resp, p, msg, req);
if (p->rtp) {
ast_rtp_offered_from_local(p->rtp, 0);
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);
@@ -9329,6 +9342,8 @@ static char *function_sippeer(struct ast_channel *chan, char *cmd, char *data, c
snprintf(buf, len, "%d", peer->call_limit);
} else if (!strcasecmp(colname, "curcalls")) {
snprintf(buf, len, "%d", peer->inUse);
} else if (!strcasecmp(colname, "accountcode")) {
ast_copy_string(buf, peer->accountcode, len);
} else if (!strcasecmp(colname, "useragent")) {
ast_copy_string(buf, peer->useragent, len);
} else if (!strcasecmp(colname, "mailbox")) {
@@ -9388,6 +9403,7 @@ struct ast_custom_function sippeer_function = {
"- curcalls Current amount of calls \n"
" Only available if call-limit is set\n"
"- language Default language for peer\n"
"- accountcode Account code for this peer\n"
"- useragent Current user agent id for peer\n"
"- codec[x] Preferred codec index number 'x' (beginning with zero).\n"
"\n"
@@ -9549,12 +9565,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
break;
case 183: /* Session progress */
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
process_sdp(p, req);
}
if (!ignore && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
if (!ignore && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
break;
case 200: /* 200 OK on invite - someone's answering our call */