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These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Steve Murphy
2006-11-07 23:46:41 +00:00
parent b3bf131dd8
commit 517978fd5f
23 changed files with 254 additions and 165 deletions
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19
channels/chan_gtalk.c
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@@ -896,8 +896,13 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
struct ast_channel *tmp;
int fmt;
int what;
char *n2;
tmp = ast_channel_alloc(1);
if (title)
n2 = title;
else
n2 = i->us;
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, "Gtalk/%s-%04lx", n2, ast_random() & 0xffff);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate Gtalk channel structure!\n");
return NULL;
@@ -916,11 +921,6 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
if (title)
ast_string_field_build(tmp, name, "Gtalk/%s-%04lx", title, ast_random() & 0xffff);
else
ast_string_field_build(tmp, name, "Gtalk/%s-%04lx", i->us, ast_random() & 0xffff);
if (i->rtp) {
tmp->fds[0] = ast_rtp_fd(i->rtp);
tmp->fds[1] = ast_rtcp_fd(i->rtp);
@@ -955,11 +955,14 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
ast_mutex_unlock(&usecnt_lock);
ast_copy_string(tmp->context, client->context, sizeof(tmp->context));
ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
ast_set_callerid(tmp, i->cid_num, i->cid_name, i->cid_num);
/* Don't use ast_set_callerid() here because it will
* generate a needless NewCallerID event */
tmp->cid.cid_num = ast_strdup(l->cid_num);
tmp->cid.cid_ani = ast_strdup(l->cid_num);
tmp->cid.cid_name = ast_strdup(l->cid_name);
if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
tmp->cid.cid_dnid = ast_strdup(i->exten);
tmp->priority = 1;
ast_setstate(tmp, state);
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {