A little look at the dialplan which rings your extension, or get dtmf, and
plays DTMF will help better understand. btw you can set the
context/extension/priority in a call file to skip some priorities of a
particular extension set.
On Fri, Sep 16, 2011 at 12:18 AM, ERIC HERRON e...@lanline.com
I have been deploying Asterisk (open source PABX) in the company which I
work.
So far, all the Asterisk servers do not really talk to each other.
Recently, I am experimenting to dial from one Asterisk server to another
through the WAN and I encountered a no-audio problem although the
callee's
The image you provided didn't open so I'm not sure about the design. If you
can send some SIP flow diagram and Asterisk CLI logs maybe it'll help
understand the problem.
On Fri, Sep 16, 2011 at 1:28 AM, Gilles codecompl...@free.fr wrote:
Hello
My ISP provides an FXS port to plug a
Am 15.09.2011 21:18, schrieb ERIC HERRON:
Asterisk 1.4.26 keeps randomly crashing then restarting itself on my
live production.
I cannot run valgrind and I do not have the right flags set in menuselect.
I can however at the dead of the night run stress tests.
I want
This obviously is pointing to NAT issue. see if you've configured both
asterisk servers with externip= PUBLICIPOFAsterisks.
Studying SIP traces on each console and specially looking at the SDPs in
INVITE will help you find out exact problem. I expect that one of the
asterisk box is sending the
On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com
wrote:
The image you provided didn't open so I'm not sure about the design.
Sorry about that. It's a PNG file and it opens in the two browsers I
tried.
The reason I don't simply get a subscription with a VoIP provider and
must go
The image just don't open for me, a wild from appears and tells me Domain
blocked bla bla. Try attaching image in this mail.
Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at
home connected to their ADSL modem so that they can make free calls
from overseas?
LOL- Its like
Hi Gilles,
Sorry about that. It's a PNG file and it opens in the two browsers I
tried.
It opens here too. It's very simple though. I would put it like this:
VOIP phone ---SIP over the internet--- Asterisk ---internal FXO
card--- PSTN-outlet ---PSTN--- PSTN phone
Can Dahdi/Asterisk do that?
Lee, John (Sydney) wrote:
I have been deploying Asterisk (open source PABX) in the company which I work.
Sofar, all the Asterisk servers do not really talk to each other. Recently, I
am experimenting to dial from one Asterisk server to another through the WAN
and I encountered a no-audio
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
I think this is a very common situation, so I'm not really sure what
your problem is. Perhaps it's because I don't use an internal card,
but in my situation it works just fine. I dial a number on my SIP
phone, Asterisk
On 09/15/2011 10:46 AM, Gustavo Santos wrote:
I understand. I'm interested in simulate the real situation because I'm
doing an academic comparative between algorithms, and is really
interesting have all possible situations.
In the real situation I use a E1 to connect a PBX through a R2 link, so
Hi,
Here (http://www.voip-info.org/wiki/view/Asterisk+database) you can read a
db1_dump185 tool can be build using asterisk source code and special make
options.
I took a look a t/usr/src/asterisk/main/db1-ast directory.
It includes a Makefile file in which db1_dump185 is present but command
Asterisk will send the two SIP endpoints 'reinvite' messages, so that they talk
RTP directly with each other. Depending on your version of Asterisk, setting
the 'canreinvite' or 'directmedia' option may make a difference, since that
will keep the traffic flowing through the servers, and the
On 09/16/2011 06:13 AM, Gilles wrote:
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
I think this is a very common situation, so I'm not really sure what
your problem is. Perhaps it's because I don't use an internal card,
but in my situation it works just fine.
Ok, I will try to do something like this. Thank you very much for the help.
2011/9/16 Kevin P. Fleming kpflem...@digium.com
On 09/15/2011 10:46 AM, Gustavo Santos wrote:
I understand. I'm interested in simulate the real situation because I'm
doing an academic comparative between algorithms,
Hello List,
I need help on disabling DTMF from a caller for a specific set of dialplan
commands and enable DTMF for some other dialplan part. This is not a SIP
peer - just live incoming call on SIP.
Please help.
Thanks
-Sammy
--
I know that I could jerry-rig something that would get me caller announce from
my Asterisk box, itself, but what I'd really like is a phone that does it like
my Panasonics. Panasonic has a beautiful DECT/SIP series of handsets... but I
guess they're aimed at the office, and jeepers, nobody wants
I would do this with ex-girlfriend logic
[mycontext]
Exten = s,1,playback(instructions)
Exten = s/5551212,n,goto(end)
Exten = s,n,read(var,prompt, .)
Exten = s,n,process..
Exten =s(end),n,hangup
From: asterisk-users-boun...@lists.digium.com
Hey :)
Smiles on your reply but Its complicated :P
Anyways I was actually using SayUNixTime() application and found out that
if a digit is pressed it breaks and go to that extension. So I wanted user
to listen to the time but key presses don't do any harm as well.
I've successfully done it now
+1 on my new Asterisk command for today.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Govind
Sent: Friday, September 16, 2011 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Fri, 16 Sep 2011 10:54:48 -0500, Kevin P. Fleming
kpflem...@digium.com wrote:
This is true, but you already answered your own question in your
original post: since Asterisk cannot know whether the called party
(dialing out via an FXO port) has answered or not, it assumes the
outgoing call is
It does on PRI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, September 16, 2011 7:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Monitoring second leg being
On Fri, 16 Sep 2011 19:35:19 -0400, Eric Wieling ewiel...@nyigc.com
wrote:
It does on PRI.
Unfortunately, this is for an ADSL modem, hence the connection to its
FXS port :-/
--
_
-- Bandwidth and Colocation Provided by
Thanks for the confirmation. Too bad Dahdi doesn't provide
call supervision so that Asterisk knows if/when the callee
has answered.
I'll experiment and see how it goes.
DAHDI with an FXO card can support call answer/hangup supervison.
Check out chan_dahdi.conf options;
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