diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-06 16:18:49 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-06 16:18:49 +0000 |
| commit | db370b01e9ecae8a6a1af55f5a23ad55ac6acd3b (patch) | |
| tree | 0d90cd3b0e8dbd13f0019ac5a46569bed8132a58 | |
| parent | 283b5f3ef054bfebf70d56016d384705622b9351 (diff) | |
Merged revisions 91439 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 lines
Add support for accepting and sending T.38 in the initial INVITE.
(closes issue #9402)
Reported by: thdei
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91440 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 18 |
1 files changed, 18 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 06a84b5c5..d92e981de 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -4723,6 +4723,10 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit if (i->rtp) ast_jb_configure(tmp, &global_jbconf); + /* If the INVITE contains T.38 SDP information set the proper channel variable so a created outgoing call will also have T.38 */ + if (i->udptl && i->t38.state == T38_PEER_DIRECT) + pbx_builtin_setvar_helper(tmp, "_T38CALL", "1"); + /* Set channel variables for this call from configuration */ for (v = i->chanvars ; v ; v = v->next) pbx_builtin_setvar_helper(tmp, v->name, v->value); @@ -13613,6 +13617,20 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru if (p->owner && !req->ignore) ast_queue_control(p->owner, AST_CONTROL_CONGESTION); p->needdestroy = 1; + } else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) { + /* We tried to send T.38 out in an initial INVITE and the remote side rejected it, + right now we can't fall back to audio so totally abort. + */ + p->t38.state = T38_DISABLED; + /* Try to reset RTP timers */ + ast_rtp_set_rtptimers_onhold(p->rtp); + ast_log(LOG_ERROR, "Got error on T.38 initial invite. Bailing out.\n"); + + /* The dialog is now terminated */ + if (p->owner && !req->ignore) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + p->needdestroy = 1; + sip_alreadygone(p); } else { /* We can't set up this call, so give up */ if (p->owner && !req->ignore) |