diff options
| author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-05-09 11:44:50 +0000 |
|---|---|---|
| committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-05-09 11:44:50 +0000 |
| commit | 3da3803c07ac5061123a9ef505c6506f07a58540 (patch) | |
| tree | d57b0a40dccb81bbcb68b1d0c7e720f6a331ffbb /apps/app_dial.c | |
| parent | faf703671d6bee04df9dc3841c30e37efe1c1c80 (diff) | |
Make SIP early media work more efficiently without so many reinvites
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26019 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'apps/app_dial.c')
| -rw-r--r-- | apps/app_dial.c | 15 |
1 files changed, 13 insertions, 2 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index 6dddb855b..7d97187e3 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -482,7 +482,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l ast_clear_flag(o, DIAL_STILLGOING); HANDLE_CAUSE(cause, in); } else { - ast_rtp_make_compatible(c, in); + ast_rtp_make_compatible(c, in, single); if (c->cid.cid_num) free(c->cid.cid_num); c->cid.cid_num = NULL; @@ -550,6 +550,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP | OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | DIAL_NOFORWARDHTML); + /* Setup early media if appropriate */ + ast_rtp_early_media(in, peer); } /* If call has been answered, then the eventual hangup is likely to be normal hangup */ in->hangupcause = AST_CAUSE_NORMAL_CLEARING; @@ -576,6 +578,9 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l case AST_CONTROL_RINGING: if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name); + /* Setup early media if appropriate */ + if (single) + ast_rtp_early_media(in, c); if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) { ast_indicate(in, AST_CONTROL_RINGING); (*sentringing)++; @@ -584,6 +589,9 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l case AST_CONTROL_PROGRESS: if (option_verbose > 2) ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name); + /* Setup early media if appropriate */ + if (single) + ast_rtp_early_media(in, c); if (!ast_test_flag(outgoing, OPT_RINGBACK)) ast_indicate(in, AST_CONTROL_PROGRESS); break; @@ -595,6 +603,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l case AST_CONTROL_PROCEEDING: if (option_verbose > 2) ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name); + if (single) + ast_rtp_early_media(in, c); if (!ast_test_flag(outgoing, OPT_RINGBACK)) ast_indicate(in, AST_CONTROL_PROCEEDING); break; @@ -1056,7 +1066,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags } /* Setup outgoing SDP to match incoming one */ - ast_rtp_make_compatible(tmp->chan, chan); + ast_rtp_make_compatible(tmp->chan, chan, !outgoing && !rest); /* Inherit specially named variables from parent channel */ ast_channel_inherit_variables(chan, tmp->chan); @@ -1550,6 +1560,7 @@ out: sentringing = 0; ast_indicate(chan, -1); } + ast_rtp_early_media(chan, NULL); hanguptree(outgoing, NULL); pbx_builtin_setvar_helper(chan, "DIALSTATUS", status); if (option_debug) |