Hello,
i using asterisk with DIVA server by CAPI termination. But when i call on
off mobile phone, i can listen normaly tone, not operator message about
availability user.
Can you explain me where are possible mistake?
Thanks
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Hi. I've just configured a PAP2 where i wanna receive faxes, but i
don't know how to disable echo-cancel option only in a misdn channel,
'cause i need it for the voice calls but not for faxes. I've just read
that it's so necessary to set it up with no echo-cancel support. isn't
it? I'm so lost
First of all, thanks a lot.
Believe me that if I'm writing down here it's due that i cannot find the
problem out. Maybe it's a bug, but either of IAX or mISDN couldn't get
pickup calls.
Could be the GrandStream?
El jue, 22-03-2007 a las 19:53 -0500, Lacy Moore - Aspendora escribió:
On
escribió:
Lukas wrote:
Hi. I've just configured a PAP2 where i wanna receive faxes, but i
don't know how to disable echo-cancel option only in a misdn channel,
'cause i need it for the voice calls but not for faxes. I've just read
that it's so necessary to set it up with no echo-cancel
, 22-03-2007 a las 21:44 -0500, Lacy Moore - Aspendora escribió:
On 3/22/07, Lukas [EMAIL PROTECTED] wrote:
Could be the GrandStream?
What's the firmware version and hardware version (if the hardware is
v2, it will say v2.0 on the back of the phone)?
You may be using older firmware
:
On 3/22/07, Lukas [EMAIL PROTECTED] wrote:
It's so strange... i don't know what's happens.
Looks like _I'm_ the one that needs to read the documentation. I
thought you could pick up a ringing Zaptel channel. But, I couldn't
get it to work on my 1.4.2 system.
I could pick up the Sip
Can you show me your extensions.conf ? I think the problem is right
there.
Something else... Spanish?
El sáb, 24-03-2007 a las 17:12 -0600, Alvaro Parres escribió:
I had set it
On 3/21/07, LKS GMAIL [EMAIL PROTECTED] wrote:
Try to set the callgroup and pickupgroup up in
Hi guys.
I have the same problem and donnow why.
I set up Nat=yes and Qulify=yes ... and notthin' happens.
So..do you think it's related to a perl function?
If u find the sol , tell us please. THANKS.
El dom, 01-04-2007 a las 18:02 +0200, Edoardo Serra escribió:
Hi guys,
I think I
Hi there!
I compiled asterisk on a HP DL380 G4 with Suse Linux Enterprise Server 9
(gcc 3.3.3). It compiled without any errors.
I also had no problems with installing my digium hardware (WC TE110P).
But when I try to start asterisk, I get the following error messages:
The error messages
Apr 12
,
because it is really complex piece of code.
The question is: is there any signaling implemented in Asterisk, which
will work with this type of CAS signaling?
Thanks for any help.
Lukas
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.
Possible alarm states (based on zaptel.h 1.2):
1. No alarms
2. Recovering from alarm
3. In loopback (local loopback or far end?)
4. Yellow Alarm (is it only Far end Loss of Frame?)
5. Red Alarm (Loss of Signal?)
6. Blue Alarm (AIS?)
7. Not Open
Thank you for any help.
Lukas Rypl
Message: 18 Date: Mon, 19 Jan 2009 19:56:14 +0200 From: Tzafrir Cohen
tzafrir.co...@xorcom.com
Subject: Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms To:
asterisk-users@lists.digium.com Message-ID:
On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote:
I am missing any
message similar to Binary content: matched instead of expected line.
It is necessary to use strings -a to filter output. So your example
should be:
asterisk -rx 'core show channels' | strings -a | grep SIP/7000
Hope it helps
Lukas
://www.voip-info.org/wiki/view/Asterisk+func+speex
and *CLI core show function AGC
Hope it helps
Lukas
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Hi there,
Thanks for the tip ! I am happily using this service now.
One question though : I cannot get DTMF to work. Is there anything I can
do in my asterisk setup to fix this ?
Thanks,
Lukas
trixter aka Bret McDanel wrote:
http://www.trxtel.com/index.php?page=Tollfree_Termination
with isdn4linux, but not the
digium cards
Lukas
Neil K wrote:
Bram,
I'm not sure this issue will be one of performance. I think you will
find cross-compiling Asterisk to the MIPS platform more challenging.
For this reason, I would recommend using the RAQ4 on the basis that at
K6-2/3 is based
absolutely on bad
place?
May it be problem with some network connectivity ? Or codec
incompatibilty ?
I saw this message only here, not on my another asterisk
installation...
Thank you very much,
Lukas
or #
are from ngrep):
=
U 2008/02/02 22:05:48.730998 192.168.1.100:5060 - 69.90.155.70:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK79892063;rport.
From: Lukas Barth sip:[EMAIL PROTECTED];tag=as2c94d5ce
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