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Code simplification

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Philippe Sultan
2008-05-28 08:39:10 +00:00
parent a06df81624
commit 2ab8f076bf
1 changed files with 40 additions and 38 deletions
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78
channels/chan_jingle.c
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@@ -956,7 +956,7 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
struct jingle_pvt *p, *tmp = client->p;
struct ast_channel *chan;
int res;
iks *payload_type;
iks *codec;
/* Make sure our new call doesn't exist yet */
while (tmp) {
@@ -974,45 +974,47 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
return -1;
}
chan = jingle_new(client, p, AST_STATE_DOWN, pak->from->user);
if (chan) {
ast_mutex_lock(&p->lock);
ast_copy_string(p->them, pak->from->full, sizeof(p->them));
if (iks_find_attrib(pak->query, JINGLE_SID)) {
ast_copy_string(p->sid, iks_find_attrib(pak->query, JINGLE_SID),
sizeof(p->sid));
}
payload_type = iks_child(iks_child(iks_child(iks_child(pak->x))));
while (payload_type) {
ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(payload_type, "id")));
ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(payload_type, "id")), "audio", iks_find_attrib(payload_type, "name"), 0);
payload_type = iks_next(payload_type);
}
ast_mutex_unlock(&p->lock);
ast_setstate(chan, AST_STATE_RING);
res = ast_pbx_start(chan);
switch (res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
jingle_response(client, pak, "service-unavailable", NULL);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
jingle_response(client, pak, "service-unavailable", NULL);
break;
case AST_PBX_SUCCESS:
jingle_response(client, pak, NULL, NULL);
jingle_create_candidates(client, p,
iks_find_attrib(pak->query, JINGLE_SID),
iks_find_attrib(pak->x, "from"));
/* nothing to do */
break;
}
} else {
if (!chan) {
jingle_free_pvt(client, p);
return -1;
}
ast_mutex_lock(&p->lock);
ast_copy_string(p->them, pak->from->full, sizeof(p->them));
if (iks_find_attrib(pak->query, JINGLE_SID)) {
ast_copy_string(p->sid, iks_find_attrib(pak->query, JINGLE_SID),
sizeof(p->sid));
}
/* codec points to the first <payload-type/> tag */
codec = iks_child(iks_child(iks_child(iks_child(pak->x))));
while (codec) {
ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
ast_mutex_unlock(&p->lock);
ast_setstate(chan, AST_STATE_RING);
res = ast_pbx_start(chan);
switch (res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
jingle_response(client, pak, "service-unavailable", NULL);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
jingle_response(client, pak, "service-unavailable", NULL);
break;
case AST_PBX_SUCCESS:
jingle_response(client, pak, NULL, NULL);
jingle_create_candidates(client, p,
iks_find_attrib(pak->query, JINGLE_SID),
iks_find_attrib(pak->x, "from"));
/* nothing to do */
break;
}
return 1;
}