Hey Ira,
I'm glad someone else noticed this. I found out this with 3-4
installs the hard way.
Latest (or should I say last X releases) zaptel 1.2 have some strange
bug on wctdm and will not working OK for incoming calls. The problem
described is exactly what I experienced, if you unload and loa
Congrats on going forward with the project Moises. MFC/R2 support on
chan_zap sounds great, looking forward on trying it out.
Regards,
Lex
On Thu, Apr 24, 2008 at 3:03 PM, Moises Silva <[EMAIL PROTECTED]> wrote:
> > Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport
> > o
Ruben,
I am also in Monterrey and have used digium hardware on R2 and PRI.
MFC/R2 is not supported by digium but the zaptel driver requirement is
the same.. what changes is using libpri vs unicall.
Just go ahead and ask them for the firmware update or as Tzafir says
use a newer zaptel that should
t; But know i have de same probelm with my incoming audio stream gets
> > clipped / dropped when you speak.
>
> Please contact Digium technical support about this. This is definitely
> something that we need to work with the vendor of the echo canceller IP
> about.
>
>
Hi,
I've used all kinds of digium cards without troubles. My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high. The
call basically feels as
007-01-07 at 18:12 -0600, Lex Lethol wrote:
> Apparently asterisk's default way to a 3-way conference lets the user
> in the middle hangup and the other parties stay on the conversation.
> This is great some times but it creates quite a bit of problems when
> trunks dont have di
Apparently asterisk's default way to a 3-way conference lets the user
in the middle hangup and the other parties stay on the conversation.
This is great some times but it creates quite a bit of problems when
trunks dont have disconnect supervision or when trying to do
accounting and billing on the
Hi yusuf,
I am working right now on a similar setup.
If its the PRI type theres not so much on the syncing part. You need
the PRI crossover rj45, theres info on voip-info on that and Orion has
software to configure via Serial cable the E1 PRI as NET/USER and Time
syncs.
I setup mine via zaptel
Hi,
Ive been playing on a asterisk to orion gsm box E1 pri setup.
I have achieved incoming calls to be passed to my asterisk box
successfully but outgoing calls will just
I have tried playing with various pridialplan and overlapdial settings
and with no success. If anyone can make more sense f
Hi,
I'm having a problem with one of our 7960. They all run latest 7.4
SIP firmware.
The problem appears on the other end. The other end constantly hears
a 'crackling' noise. I have tested using phone set, headset and
speaker and the noise appears on all cases. I have other 7960 setup
exactly
the phone mentioned here?
>
> http://voxilla.com/voxstory134.html
>
>
> _
> Mobilcom
> http://www.mobilcom.net
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol
> Sent: T
Hi,
Just finished watching the season finale of '24' the TV series.
Throughout the series they have been showcasing Cisco hardware
especially Cisco IP phones (7970's).
On the last episode or two they showed what seemed to me a new cisco
IP video phone. It stands just as a 12" lcd screen with th
Ive been using it too and its working great. Still waiting for my DID
but as far as terminating to the US I am very impressed with sound
quality.
Lethol
On Mon, 13 Dec 2004 16:51:47 -0800 (PST), Steve Edwards
<[EMAIL PROTECTED]> wrote:
> I've used them for a couple of months. My usage is very sm
So, if zaptel will not read codes from my indications conf file, what
would be a suitable solution to feed it my country tones?
Is there a list of loadzone/defaultzone country codes included in
zonedata.c? Can this list be updated to include my country? (I have
the tones)
Thanks for the help
Let
Does anyone know if this needs any special modification to work
outside the US? I have setup my country's correct tone info and
tested thru the indication.conf file.
Question would be, where does my zaptel device get the tones expected
for the busydetect procedure? How can I modify them? Is this
I am getting a system hang ups as well.. But my system will halt after
being about 10 minutes on a call.. and there is no error showing up on
asterisk CLI :S My linux server will just freeze and will only happen
while on asterisk
I have no ida on how to debug this one. I think it might ne a
hard
Im also interested in a couple of these... plesase email me if you are
selling or post over a link!
Lethol
On Fri, 24 Sep 2004 22:02:52 -0400, William Suffill
<[EMAIL PROTECTED]> wrote:
> Anyone here have any pointers of where to get 1 of the PAP2-NA. Given
> all the talk about it I'd be curious
<[EMAIL PROTECTED]> wrote:
> To be Slashdotted within 30 minutes.
>
> -Ken Shaw...
>
> On Thu, 2004-09-23 at 07:28, Lex Lethol wrote:
> > Hi,
> >
> > Reporting from Astricon, Mark uploaded the 1.0 release while giving
> > his speech a few mintues ago..
>
I tried the xten one and didn;t like at all..
Havent tried to SJPhone, but my guess is that it has better support.
Lethol
On Thu, 23 Sep 2004 08:13:10 +0200 (CEST), Peter Svensson
<[EMAIL PROTECTED]> wrote:
> On 22 Sep 2004, Sudhir Kumar wrote:
>
> > Is there a soft phone for PocketPC or iPaq?
Hi,
Reporting from Astricon, Mark uploaded the 1.0 release while giving
his speech a few mintues ago..
Bring out the champagne :)
Lethol
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBS
Stay away from the 7910 if your going SIP. It will not support it.
Lethol
On Thu, 23 Sep 2004 01:45:23 +1000, Shaun Ewing <[EMAIL PROTECTED]> wrote:
> The 7910 does not support SIP. It is SCCP only.
>
> -Shaun
>
>
>
>
> - Original Message -
> From: Henry Devito <[EMAIL PROTECTED]>
hi Flynn,
I have an OpenLine4 on my setup. Everything appears to work finw and
I am not having the hangup detect but I am having problems when
voicemail tries to record via vpb channel. Did you ever have that on
your OpenLine4?
I have not tried out the OpenSwitch12 but I am a bit scared with
vo
Benjk,
I dont have an answer to your problem, but I am currently using the
same asterisk CVS HEAD found in voicetronix webpage. Most features
are working OK and I am currently trying fo fix a voicemail problem
but it appears not to be related to loopdrop. Are you sure the card
works fine? (hardw
Heya Kelvin,
Are you using the latest asterisk download from voicetronix webpage.
I got most asterisk features working with an OpenLine4 but I still
have some bugs/incompatibility issues to resolve.
Make sure you download the latest driver and asterisk and make. After
installing the voicetronix
It definitely sounded sarcastic :P
Lethol
On Mon, 30 Aug 2004 08:21:06 -0400, Leif Madsen <[EMAIL PROTECTED]> wrote:
> On Mon, 30 Aug 2004 10:21:55 +0800, Joseph Shi <[EMAIL PROTECTED]> wrote:
> > Steve Underwood Wrote:
> > >Just wait for the simplified Chinese version to appear in Shenzhen's
> >
Hi,
I am fairly new to asterisk. I am currently testing my first setup.
I've been able to debug most of the problems to make asterisk work
with my hardware setup until this time.
Currently I have the following issue:
Voicemail is running but when I test to leave a voicemail thru my
incoming PST
On my experience, you should go to SIP whenever possible. 7940/60 on
SIP will do most if not all fuctions.
Try the little chart on support hardware on chan-sccp.sourceforge.net
Lethol
- Original Message -
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
Date: Fri, 27 Aug 2004 14:16:11 +010
Hi,
I am using a voicetronix OpenLine4. I downloaded a recent asterisk
CVS from voicetronix webpage but have had no luck to reduce echo on
outgoing calls and for it not to crach on incoming calls. I dont
think both problems are related though.
Here is an output of what happens when a new call c
Make sure not to buy any 7910 if you want an all SIP network. I dont
see any advantages for it specially if you are in a planning stage.
I also have seen tons of posts sayin that the 7920's are a pain to get
'em working.
Lethol
On Wed, 18 Aug 2004 17:06:17 -0700, Scott Laird <[EMAIL PROTECTED]>
Julien,
Just to let you know that I manually included your patch and
everything compiled OK.
I'll begin testing now.
Thanks!
Lethol
On Tue, 17 Aug 2004 02:02:58 -0600, Lex Lethol <[EMAIL PROTECTED]> wrote:
> Hi Julien,
>
> Thanks for the feedback. I am currently trying
wrote:
> On Mon, Aug 16, 2004 at 04:46:56PM -0500, Lex Lethol arranged a set of bits into the
> following:
> > I recently bought a 7910. I found out too late that it would not do
> > SIP as I initially thought. Anyway before ditchingit for a 7960 I
> > wanted to try it ou
Hi,
I recently bought a 7910. I found out too late that it would not do
SIP as I initially thought. Anyway before ditchingit for a 7960 I
wanted to try it out, I read that the guys at
http://chan-sccp.sourceforge.net/ had done some improvements to the
original chan_sccp driver and having 80% fun
Hi,
I recently bought a 7910. I found out too late that it would not do
SIP as I initially thought. Anyway before ditchingit for a 7960 I
wanted to try it out, I read that the guys at
http://chan-sccp.sourceforge.net/ had done some improvements to the
original chan_sccp driver and having 80% fu
33 matches
Mail list logo
