Hi people!
I am not getting smart getting asterisk 1.6 behind a NAT to run.
1. I enabled IP forwarding on debian linux
2. told asterisk in general that he is behind NAT and mentioned him
his external static IP Adress as well his domain in the outside world.
If a client who is connected with a
Hi people!
Do you guys know where to find a STUN Server Howto?! Why?! We all know,
to get Asterisk behind an NAT Router to run, is a bit tricky, and you
might have to fire a lot of holes in your firewall.
However, I would appreciate it very much if somebody could give me great
links of how to
Hi people!
Asterisk PBX (version 1.6.5): I have Asterisk behind a NAT (192.168.1.2)
SIP Phone: A client behind NAT (192.168.1.3)
Softphone: One other client somewhere in the internet (also behind an NAT).
they want to speak with each other, and if they do, there is no sound.
if softphone in the
Hi people!
I am not getting really smart. I get the SVN Edition of asterisk GUI
interface, compiled and love to get it to run, what won't work. What am
I doing wrong?!
svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0
make
make checkconfig
make install
and If I open one of the
Hi people!
I am not getting really smart. I get the SVN Edition of asterisk GUI
interface, compiled and love to get it to run, what won't work. What am
I doing wrong?!
svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0
make
make checkconfig
make install
and If I open one of the
Hi!
I have problems building asterisk 1.6.0.6.
./configure --prefix=/usr
make
gets me:
enerating embedded module rules ...
[CC] extconf.c - extconf.o
In file included from /usr/local/include/datatypes.h:50,
from /usr/local/include/err.h:49,
from
install to /usr/local or even some special place
just for * just to help keep the box more organized.
--
Matt
On Mon, Feb 23, 2009 at 7:17 PM, Tamer Higazi th9...@googlemail.com
mailto:th9...@googlemail.com wrote:
Hi!
I have problems building asterisk 1.6.0.6.
./configure
= /usr/local2 what
sounds for me more then strange (to take a whole partition out.
Tamer
Tzafrir Cohen schrieb:
On Tue, Feb 24, 2009 at 09:31:21AM +0100, Tamer Higazi wrote:
I did the same thing, without the prefix stuff!
The same error!
[CC] extconf.c - extconf.o
In file included from
Hi people!
I want to save the 1st letter from the ${EXTEN} variable. I don't want
to trim it, I want to RESAVE it into a new variable.
Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
I would thank you for all advises.
Tamer
})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] receiving 1st
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receiving 1st digit from a variable
exten = _[0-1]X.,1,Set(MSNCHOICE=${EXTEN,1:1})
brings me
Hi people!
I need help according getting asterisk 1.6.0.6 installed. I posted to
digium, but it seems to be that it is not an error, but either I am not
getting smart what I have to do, to get it solved (configured and
installed as well).
./configure
make
gets me this output:
In file included
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: domingo, 22 de marzo de 2009 05:59 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] make script 1.6.0.6 breaks up, need help!
Hi people!
I need help according
Hi people!
I am coding a special sollution for that I need to know if I can send
AT commands in the extensions.conf, to one subscriber. Is there a way
doing this through asterisk 1.6 ?! For sure anybody of you, would as
why I want to do that. I want to speak to my endsystem directly with
AT
Hi!
I have a Grandstream VoIP Device, at which a DECT base with 2 cordless
phones are connected. If a call is placed and made through one cordless
phone the other cordless phone appears as busy.
What I want:
1. The Base station of the DECT cordless phones, is connected at 1 FXS
Port of my
-Commercial Discussion
Subject: Re: [asterisk-users] opening 2 and more channels on 1 SIP account
On Fri, 17 Apr 2009, Tamer Higazi wrote:
Sorry I write this message into the developer lailing list, I do this
because nobody in the user list could answer me this question, due it's
to technical
that gives you the chance
to make a lot! It is programming work, not more then that.
I hope all questions are being answered.
Tamer
Steve Edwards schrieb:
On Sat, 18 Apr 2009, Tamer Higazi wrote:
On Fri, 17 Apr 2009, Steve Edwards wrote:
Impatient are we :)
Yes you are entirely
D Tucny schrieb:
2009/4/18 Tamer Higazi th9...@googlemail.com
mailto:th9...@googlemail.com
Scenario:
I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP
cpu to take out the echo cancellation.
Communication is done through the chan_capi interface module
Hi people!
I am looking for a h.323 implementation guide for asterisk. I looked in
the doc folder of the latest asterisk source distribution and I didn't
fund anything acording to this subject.
If you guys could give me any advise, I would thank you.
Tamer
after the tone and hang
up or press the pound key.
Is there a way to delete the second part from the voicemail, that only
my personal recorded message is played back and a signal tone comes to
signal the caller to start talking?!
Tamer Higazi
Hi people,
I have a Cisco ATA 186 which understands only the SCCP protocoll,
therefore I am a pure beginner and I hope that anybody of you could help
me.
How will I configure the ATA which has 2 analog ports?
For any support I would kindly thank you
Tamer
--
Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
are registered, one in TE and the other on in NT mode.
Calls do successfully come inside, but I want to connect my ISDN phone
at the board, but
make the power supply for the phone
somehow?!
Tamer
Am 30.08.2011 20:05, schrieb Patrick Lists:
On 08/30/2011 06:32 PM, Tamer Higazi wrote:
Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
Do you want to run the entire PBX on the Android client or are you just
looking for a IAX programm to be installed for receiving calls?!
I think this is what you ment.
Here is the url:
https://market.android.com/details?id=com.bw.iax.ui
Am 02.09.2011 16:32, schrieb Gilles:
On Fri, 2 Sep 2011
I advise you taking Gentoo Linux. There is a great asterisk repisetory.
Also support patvhes for NON digium hardware.
Resources:
http://oreilly.com/openbook/
there you find on the right the free book Asterisk: The Future of telephony
the 3rd edition is available, but that book covers every
what do you mean exactly?! One what?! What do you plan to accomplish?!
Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards
are really expensive, not under 400.- € inkluding DSP Processors.
I advise you taking Gentoo Linux, getting asterisk on it and put a
single Port HFC-S
Am 20.09.2011 19:47, schrieb Gopal krishnan:
What is the difference between using mISDN for BRI and using Dahdi
mISDN was at 1st done for ISDN Services and channel driver as I know. It
supported like call routing (switch based, not your side on the pbx level).
without mISDN?
you can use
for your comments, really your comments are useful. And
finally I think using dahdi instead of mISDN is better.
On Wed, Sep 21, 2011 at 3:10 AM, Tamer Higazi th9...@googlemail.com
mailto:th9...@googlemail.com wrote:
Am 20.09.2011 19:47, schrieb Gopal krishnan:
What is the difference
2.4.x does it with me, so I am sure 2.5.x do makes it either!
Tamer
Am 21.09.2011 16:13, schrieb Olivier:
Hello,
Is Dahdi 2.5.0 supposed to support BRI NT PtmP mode ?
Regards
--
_
-- Bandwidth and Colocation
compare the prices between sangoma and digium pri boards!
Sangoma's oards here in Germany are cheaper as the ones from digium.
if you need detailed help, you can contact me, and I can workout for you
something as well as helping you setting up your pbx!
Tamer
Am 23.09.2011 15:01, schrieb
Go to ATT and ask for a BRI ISDN Line. Tell them that you want a
system access with a number block.
that would be typically 54448[0-9]
where your extensions are from 0-9
I don't know what protocol the americans are using, as I know the
americans made for sure their own thing which is known as
permitted to order BRIs.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Friday, September 30, 2011 9:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Hi!
I am interisted to dial out from the console with chan_alsa. Can
somebody of you help me according this problem?!
I added user the asterisk to pulse and pulse-access, and it didn't
change anything. alsa applications are routed by default to pulse.
cat /etc/asound.conf
pcm.!default {
type
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