Il 29/09/2014 15:57, Tzafrir Cohen ha scritto:
> On Mon, Sep 29, 2014 at 03:52:25PM +0200, Claudio ML wrote:
>> Hi,
>>
>> It's the first time i try to configure an ISDN card with dahdi, so my
>> experience is very poor (be kind ;))
>>
>> My problem is
Hi,
It's the first time i try to configure an ISDN card with dahdi, so my
experience is very poor (be kind ;))
My problem is with dahdi_genconf, when i start it it says:
/usr/sbin/dahdi_span_assignments: Missing
'/sys/bus/dahdi_devices/devices' (DAHDI driver unloaded?)
Command failed (status=256
Hi to all,
I am searching to make work an Asterisk, with an ISDN card with Cologne
Chipset.
Here is the lspci:
01:09.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
Subsystem: Cologne Chip Designs GmbH ISDN Board
Flags: bus master, mediu
Hi,
I used to install asterisk on debian squeeze with digium repository.
The last build of asterisk available is 1.8.11.1.
Is this repository discontinued ?
Thanks.
--
_
-- Bandwidth and Colocation Provided by http://www.api-di
Hi,
I have to make an asterisk gateway in front of several other asterisk.
This gateway will essentialy be used for outbound call.
This gateway will be connected to other asterisk by IAX trunk, outbound
call will use SIP trunk (voip provider or patton isdn).
I have a TE220BF available than i ca
Hi
I'm trying to install dahdi. I just need the dahdi timer for
conference.
I currently using digium debian package for asterisk 1.8.8.1.
When i install asterisk-dahdi , i've got several dependencies which
came for official debian repository (including the dahdi package) and
are outdated.
Is
Hi all,
I've been trying to get SMS operational on my Asterisk box, which has a
TDM400P card with a pair of FXO interfaces configured (ZAP/1 & ZAP/2).
I've not had luck with either of my lines, after issuing the command
"smsq --motx-channel=ZAP/1/1709400X 0 register". I see the
following out
Hello,
I hope, somone can help me.
When I try to register an ata (sipura 2002 for example), it is successfull, when
a device is installed since a few weeks on the asterisk.
It isn´t successfull, when it is a completly new device, I added to the
asterisk.
Have someone any idea?
Many thanks!
Has anyone had good success with the IAXy? I've tried everything
including PAT on the IAX2 port to the IAXy device to no avail (using the
alternate server parameter). I guess a call to Digium is in order!
Regards,
--- Gavin Adams
Promisant (USA) Inc.
O: 770-913-3727 F: 770-913-3726
__
Hi,
I've provisioned an IAXy adapter on a network segment local to my
asterisk server. Provisioning is fine, as is the registration and use of
said device. Since the local address is private address space, I setup
the public IP address of my Asterisk server as the alternate. When
taking it to anot
Will Stowe
Systems Administrator
Promisant (USA) Inc.
email:[EMAIL PROTECTED]
Office: (770) 913-3723
Mobile: (404) 993-0526
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To UNSU
testing
Will Stowe
Systems Administrator
Promisant (USA) Inc.
email:[EMAIL PROTECTED]
Office: (770) 913-3723
Mobile: (404) 993-0526
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Or you can try getting it at:
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-1.0rc1.zip
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-lin-1.0rc1.tar
:)
>
> http://iaxclient.sourceforge.net/iaxcomm-win-1.0rc1.zip
> http://iaxclient.sourceforge.net/iaxcomm-lin-1.0rc1.tar
_
Hello,
This is David Deutsch, and I’m the owner of VoiceConduits.
There seems to be some confusion related to our company, regarding the past few
posts.
VoiceConduits is currently NOT open for public business, we
have never to date advertised or attempted to attract business. It app
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
>
> I just upgraded to RC1 from a two-three month old CVS , and noticed
that
> during IAX2 calls to my service provider there are periods (usually
less
> than 10 second
> What I'm looking for is the ability to determine whether or not a queue
> has
> any queue handlers (active agents), and if it does not, bypass sending
> the
> caller to the queue and pass them on to a message or IVR system.
>
> -Chris
http://bugs.digium.com/bug_view_page.php?bug_id=214
T
Hi. I have posted a fix for announce so that it does not stop the music on hold until
after playing the
announcement file. If you can, please test it out for me.
Thanks,
Kevin
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Hi. I'm looking for simple residential Asterisk use.
I will need a local DID number in the Phoenix, AZ area which includes the area codes
480, 602, 623.
Can anyone provide termination in those area codes with SIP or IAX? I suppose H.323
is also ok, but I haven't messed with it.
Is a voicem
> > I got 1.0.4.38 from SIPphone's server at 130.94.123.253 but last time
> I
> > tried it was offering 1.0.4.35. For me 1.0.4.38 cleared all my
> problems
> > but from SIPphone's email it had hosed some phones, such that they
> were
> > talking about replacement units.
>
> I'm still on fscking
> I am experiencing a problem that from list archive it appears others are
>
> running into. When I dial from Cisco 7960 via the * to Free World
> Dialup
> destinations that supports G.729 the call fails. The major error from
> the debug log is
>
> Jan 15 00:11:14 NOTICE[22545]: channel.c:1481
Hi. Thank you to Olle, the Wiki, and Trollphone. I found the Trollphone Rate Engine
listed on the Wiki at:
http://www.voip-info.org/wiki-Asterisk+addon+rate-engine
I'm not familiar with LCR yet, but this is something that I need to do. I have it all
installed but am not familiar with the term
> The wiki says this about the MusicOnHold command:
>
> "Plays hold music specified by class. If omitted, the default music
> source for the channel will be used."
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold
>
> How do I set the default music on hold class for the
> Hi,
>
> Do the callers in USA dialling from USA Telco lines always have to
> prefix the CITY/AREA code with "1" in order
> To successfully make a call to other USA destinations?
>
>
> I have not been to USA (yet) :)
>
> Ta
> SJ
In all cases of long distance, 1 plus the area code is used
> Original Message
> Subject: RE: [Asterisk-Users] SIP and error talking to voicemail
> From: "Dave Cotton" <[EMAIL PROTECTED]>
> Date: Fri, January 09, 2004 1:03 am
> To: "Asterisk List" <[EMAIL PROTECTED]>
>
> On Fri, 2004-01-09 at 06:37, [EMAIL PROTECTED] wrote:
>
> > How co
> 130.94.123.253 came from SIPphone not Grandstream, but even
> http://www.grandstream.com/TEMP/FIRMWARE/ only has 1.0.4.30
>
> The only thing I can say is it's cleared my problems, making my GS
> usable again.
> --
> Dave Cotton <[EMAIL PROTECTED]>
How come every time I try connecting to their
On the config webpage, its on the bottom.
Kevin
> Original Message
> Subject: Re: [Asterisk-Users] Re: Grandstream Early Dial
> From: "Aaron Martin" <[EMAIL PROTECTED]>
> Date: Sun, January 04, 2004 3:49 pm
> To: [EMAIL PROTECTED]
>
> Where / how do I set DTMF payload type to 1
Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable
is not defined, I will get a parse error. Yeah, there are ways around it, but I would
think that it should return false if 0, null, or undefined. I would change it, but I
have no idea about bison and I only h
>> > Brian West wrote:
> > >
> > > its
> > >
> > > #include filename.conf
> > >
> >
> > Does the synatx include the # at the beginning of the line ?
> > And can this type of include be time/date dependant like the
> standard
> include ?
> >
> > include => filename.conf
> >
>
> Check here:
> ht
> > Lubomir Christov <[EMAIL PROTECTED]> said:
> >> Yes, I know that the Grandstream firmware have problems (I have here
> 15
> >> phones with some beta version already installed :( and waiting for
> bug
> >> fixing in the new beta) but the stable version 1.0.3.81 is working
> just
> >> perfect.
>
> >I have a call generation script (very simple) to generate
> >call load for testing, if that's what you're trying to accomplish.
> It's
> >good for generating huge call volumes for IVR testing.
> >Let me know if you need it!
I would be interested in the script. If it is small, maybe you could
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