I use Free PBX with Visual Dialplan.
Free PBX to configure extensions, trunks etc., and Visual Dialplan to
create and manage IVR.
It works perfectly for me.
Best,
Rayan
On 2/28/2013 2:47 AM, Nguye^~n Công wrote:
I have a question about management tool for asterisk. What tool are
you used
I've found a workaround of sorts, If I change my below code to :
1AA = {
NoOp(${CALLERID(num)});
Answer(); // --- add this
Ringing;
Set(CHANNEL(musicclass)=none);
Dial(${OUTBOUND-TRUNKR}/1XX,30);
Voicemail(198,u);
When Answer fixes the issue, the root cause is often NAT (could be firewall)
since Answering the call prevents any reinvites.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Friday, March 01,
We are having a weird problem where calls get cut off in the middle. I'm not a
SIP expert but could the INVITE with an empty SDP be the problem?
|Time | 209.220.119.18|
| | | 208.88.61.150 |
|9687.369 | INVITE
Kevin,
Thanks for the response.
We are on the same page..
The only problem that we have with the penalties is that a call can get 'stuck'
on an agent in this scenario.
If they walk away and forget to log off, or go DND than the call will ring with
them over and over and not move onto the
I thought it was the re-invites too, but I have it turned off everywhere.
On 03/01/13 08:36, Eric Wieling wrote:
When Answer fixes the issue, the root cause is often NAT (could be firewall)
since Answering the call prevents any reinvites.
-Original Message-
From:
I think a simple tcpdump of the traffic will show the mystery. It can be
your provider doing something nasty. Have you tried using some other cheap
SIP termination? or arrange a fake termination yourself on another server?
Leandro
2013/3/1 Gerard gsara...@rarcoa.com
I thought it was the
