diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-06 16:14:26 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-06 16:14:26 +0000 |
| commit | 643ba32cb79c2645fde8a50dc665df96cf6c9a0f (patch) | |
| tree | 05cdf1ac103878ee3139192d97a262f4f4a67676 /channels | |
| parent | 143bc3f5c3591f6f4bc00d0a505212eb40405cfd (diff) | |
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/branches/1.4@91439 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -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 31d1e3266..0407b2771 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -4052,6 +4052,10 @@ 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); @@ -12175,6 +12179,20 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) ast_queue_control(p->owner, AST_CONTROL_CONGESTION); ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + } 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 && !ast_test_flag(req, SIP_PKT_IGNORE)) + ast_queue_control(p->owner, AST_CONTROL_CONGESTION); + ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); + sip_alreadygone(p); } else { /* We can't set up this call, so give up */ if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) |