Thank you francois!
Where could you find that info ?
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de F6HQZ
Envoyé : mercredi 23 décembre 2009 22:46
À : Asterisk Users Mailing List - Non-Commercial
Hi All,
Is someone implemented Tel uri support in the latest asterisk ? If yes, can
you guys share some info on it
Regards,
Ramananda AS
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Hi,
I have problem with X100P clone card.I can not force it to work
under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2.
I looked over and over on configuration and could not see any mistakes.
Here are relevant configuration files.
/etc/dahdi/system.conf
fxsks=1
echocanceller=mg2,1
loadzone
I'm not familiar with cdr_radius, but is there a debugging option? Anything
in /var/log/messages?
On Thu, Dec 24, 2009 at 1:35 AM, Zhang Shukun bit...@gmail.com wrote:
Thank you !
i have load cdr_radius.so successfully! but another error occur.
-- Executing [4...@tutorial:1]
Ah ! It's a jamming of Digium FFA user manual, ideas and tests from my
customers and myself.
From Digium's side you can/must acces to this WEB page :
https://www.digium.com/en/supportcenter/documentation/viewdocs/FAX
I love to check Digium's solutions and to know how to use them.
So, I have
super quick asterisk in gdb howto:
compile asterisk with DONT_OPTIMIZE (in make menuconfig - Compiler flags)
gdb asterisk
run -cvv
wait for the crash
bt
bt full
and now make the patch :)
Kristijan
2009/12/24 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Dec 24, 2009 at 12:13:55PM
On Thu, Dec 24, 2009 at 01:12:58PM +0100, Kristijan Vrban wrote:
super quick asterisk in gdb howto:
compile asterisk with DONT_OPTIMIZE (in make menuconfig - Compiler flags)
which changes the behaviour of your code. Rebuilding is not always an
option.
If using Asterisk from a package, be
Hello users,
i have been testing the DTMF tone detection using originate command
both from Asterisk CLI and java API.
but my DTMF entry at the originate user is not getting detected by the
asterisk
in both the cases
what i should do to make it work
any help will be appreciated.
my versions
Whether you can do this and how successfully depends on your Asterisk
Flavor. Meetme has no internal limitation that would allow you to limit the
meetme to two callers. IMO you would have better luck using a bank of
pre-defined meetme rooms, but you can set up as many as you want on the
fly from
Barry Fawthrop wrote:
Hi All
I have been looking around and haven not been able to find a working example
I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
1.4.10.2
I use a sangoma A200 card so I am using wanpipe 3.4.7
If I use zaptel which I read I need for
Administrator TOOTAI a écrit :
Hi,
I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip
extension definition, when I set language, it is not reported in the
extensions_custom.conf file (eg language=xx).
Am I missing something or is it not the right way to set language?
Barry Fawthrop escribió:
Barry Fawthrop wrote:
Hi All
I have been looking around and haven not been able to find a working example
I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
1.4.10.2
I use a sangoma A200 card so I am using wanpipe 3.4.7
If I use zaptel
Miguel Molina wrote:
Please correct me if I'm wrong, but AFAIK spandsp based fax
applications for asterisk only support a maximum of 9600bps.
No. V.17 (speeds up to 14400 bps) are supported.
Thanks,
Lee.
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Hi Guys,
Merry Christmas and Happy new Year.
I am looking for some assistance from the group as i think this might
already have been tried before.
i have an asterisk server with a external USB Harddisk Drive, just to store
recordings. I am using the mixmonitor application for doing the
Unfortunately, sip show peers did not work in my case. The sip peers were
apparently online and OK (I use qualify=yes) but they weren't...
The SIP clients could NOT register, so they were offline but sip show peers
stated that they were OK.
I would prefer to perform an automated SIP
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you don't need to mess it up by introducing
an easy point of failure.
_
From:
It looks to me like calls from your Dial will route back to the sip-outgoing
context and Dial again... it's loop. You'd really need to provide more
logging information to advise further.
On Thu, Dec 24, 2009 at 5:16 AM, hadi motamedi motamed...@gmail.com wrote:
On Wed, Dec 23, 2009 at 1:55
sip show registry might be more helpful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Thursday, December 24, 2009 10:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] how to
Thanks but sip show registry yields nothing.
--- On Thu, 12/24/09, Danny Nicholas da...@debsinc.com wrote:
sip show registry might be more
helpful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of
On Thu, Dec 24, 2009 at 11:24:24AM -0500, Krishna Sumanth Chava wrote:
Hi Guys,
Merry Christmas and Happy new Year.
I am looking for some assistance from the group as i think this might
already have been tried before.
i have an asterisk server with a external USB Harddisk Drive, just to
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you don't need to mess it up by introducing
an easy point of
Hello,
Thanks for the reply. I am in full control of the meetme rooms since I
initiate the call for both parties and I can do next call into a new meetme
room.
Can anyone please share their AMI, PHP, and dialplan code relating to
creating MeetME rooms on the go?
Much appreciated.
Thanks
On
It looks like whatever is being transmitted, or the response, isn't getting
through. Possibly due to NAT or a firewall? It would help if you described
the scenario where this is occurring.
On Thu, Dec 24, 2009 at 7:18 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Hi,
How
On Thu, 2009-12-24 at 18:53 +0100, Gergo Csibra wrote:
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you
Dave Wrote:
It looks like whatever is being transmitted, or the response, isn't
getting through. Possibly due to NAT or a firewall? It would help if you
described the scenario where this is occurring.
Indeed, my post was gibberish :-O
This was a 'nat' issue, but not in the traditional sense.
On Wednesday 23 December 2009 12:52:38 Ira wrote:
Someone posted a message suggesting someone try sendtext() and so I
thought I'd see if it was useful. Much searching through help at the
CLI has failed to find any help for sendtext, but I did find that:
core show function vmcount fails but:
I wrote a script to check clients and restart asterisk if registrations died
(external IAX)...but you could modify for your needs. Check it out on
www.generationd.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hello,
Please forgive me if I'm repeating this post. I have searched and looked for
similar problem with a solution but have not see a similar one.
My outgoing SIP and other channels work fine but the incoming/inbound SIP
call goes straight to Broadvoice voicemail. I see that Broadvoice is
Hi all,
I am new to Asterik.
Ia m trying to compile the source with the latest asterisk-1.6.2.0 these are
the issues I am getting.
initially,I got
mkdir: cannot create directory `/var/lib/asterisk'
than after reading the archives:
I did:
./configure --enable-dev-mode --prefix=/tmp/asterisk
- Qurba Joog qurbaj...@gmail.com wrote:
| Hello,
|
| Please forgive me if I'm repeating this post. I have searched and looked for
similar problem with a solution but have not see a similar one.
|
| My outgoing SIP and other channels work fine but the incoming/inbound SIP
call goes
On Thu, Dec 24, 2009 at 10:50:01PM -0800, Aditya Kumar wrote:
Hi all,
I am new to Asterik.
Ia m trying to compile the source with the latest asterisk-1.6.2.0 these are
the issues I am getting.
initially,I got
mkdir: cannot create directory `/var/lib/asterisk'
than after reading the
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