Gurus, I am missing something with Adit 600 channel back.
- FXS cards designed for connecting phonesets or telephony lines?
Thanks for your advice
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Thank you.
Does anyone by any chance have Adit 600 FXO spare modules to sell?
Greg Smith wrote:
FXS is for stations
FXO is for exchange lines
cheers
greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vasyl
Rublyov
Sent: Monday, 23 August 2004 9:39 PM
Spectro,
This configuration works fine to me. I have Merlin Legend with 2 DS1
100D cards - one goes to upstream PRI (Verizon) and the second one goes
to Asterisk with T100P card. The timing is going from Verizon and
asterisk configured as pri_net.
Crossover cable used between Merling Legend
Are you going to use single E1 line? How many concurrent calls.
Single Xeon 3.0 with *1 GB *ram should serve 30-32 calls with not
problem, even during G729 transcoding and echo cancellation.
If you are not plaining to do G729 transcoding - then I believe you can
put more then 32 calls in this
Steven,
I just would like to apologies for my comment...
Steven Critchfield wrote:
On Thu, 2004-07-29 at 14:33, Vasyl Rublyov wrote:
I really upset from this forum... from last ten posts nothing came
back reasonable.
None for PRI problem, none for random disconnects
I'm sorry free
that might help dignose the problem.
Regards,
Michael East
- Original Message -
From: Vasyl Rublyov
To: [EMAIL PROTECTED]
Sent: Thursday, July 29, 2004 8:33 PM
Subject: Re: [Asterisk-Users] zt_pri_error: PRI: Warning:
unknown/inappropriate protocol discriminator received (00/0)
I really upset from
I really upset from this forum... from last ten posts nothing came back
reasonable.
None for PRI problem, none for random disconnects
Vasyl Rublyov wrote:
Thank you.
It works to me as well.. but some time I am getting (PRI side only)
huge noise.
reseaux wrote:
Dear
Anyone can comment this or just mailing is dead?
Vasyl Rublyov wrote:
I started to see this problem as soon as we connected to Verizon PRI
(DMS-100 Switch) and it prints every 3-5 seconds.
[Verizon DMS-100 PRI] [Lucent Merlin Legend] [Asterisk]
Asterisk/LibPRI/Zaptel are built from
, Vasyl Rublyov wrote:
Anyone can comment this or just mailing is dead?
Vasyl Rublyov wrote:
I started to see this problem as soon as we connected to Verizon PRI
(DMS-100 Switch) and it prints every 3-5 seconds.
[Verizon DMS-100 PRI] [Lucent Merlin Legend
I started to see this problem as soon as we connected to Verizon PRI
(DMS-100 Switch) and it prints every 3-5 seconds.
[Verizon DMS-100 PRI] [Lucent Merlin Legend] [Asterisk]
Asterisk/LibPRI/Zaptel are built from HEAD CVS on Jul 10 2004.
Any help?
Jul 27 20:50:20 WARNING[98310]:
.
The phone keeps rebooting and reconfiguring IP address.
What could be the problem?
Vasyl Rublyov wrote:
Can anyone tell me if this phone supports NAT ? If so then how?
Ken Wiesner wrote:
Vasyl,
Not sure what kind of setup you're trying to do but if its a build out
al Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Vasyl Rublyov
Sent: Friday, July 02, 2004 7:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Inter-Tel Eclipse2 (IP PhonePlus)
Thank you, Ken.
I just asked because one from our clients is using this sys
that absolutely screamed from att. Never again will I use
Sprint for an IP connection. They oversell many times their bandwidth and
the service sucks if there is a problem.
Anton
- Original Message -
From: "Vasyl Rublyov" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 13, 200
:( Just getting silence Is this mailing list alive at all?
Vasyl Rublyov wrote:
All,
I seen already threads about one way audio... but never seen anyone
answered completely on it.
There is a problem, what we are getting, even with stable-1, CVS
updates in May, June as well as last Saturday
Joseph,
Already tried this...
And more interesting that this works fine for days/weeks... and
randomly start giving this problems for hours/days... after that goes
away.
Even without restarting asterisk
Joseph wrote:
On Tue, 2004-07-13 at 11:50, Vasyl Rublyov wrote:
:( Just
No One Answered... so decided to go with ATT and their CoS
Vasyl Rublyov wrote:
Hello All,
We are choosing right now good Internet provider in Eastern Cost
(Washington DC metro) for our voice communication up to India
(Bangalore/VSNL) and Ukraine (UTEL).
They are both can provide to us dedicated
All,
I seen already threads about one way audio... but never seen anyone
answered completely on it.
There is a problem, what we are getting, even with stable-1, CVS updates
in May, June as well as last Saturday (Jul 10, 2004)
[T1/PRI PSTN] == [Lucent Legend PBX] == [T1/PRI] == [T100P
Asterisk
Thank you Ken.
Ken Wiesner wrote:
Jason,
We've got the Axxess and we just have the IPC (8 channel) card with the IP PhonePlus keysets. You only need to have the IPC for this type of configuration. It works, but again, not as well as one would hope.
The SIP gateway is only required if
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards,
Vasyl Rublyov
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. Not kidding,
there is a calendar in the office counting down the days till the lease is up and we can make a full migration to Asterisk.
:-)
That said... RUN!
Ken
P.s. have a good weekend all!
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vasyl Rublyov
Sent
g), paging and a few others.
Anyways, we've tried both and 2 works out best for us. Hope this info helps!
~ken
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Vasyl Rublyov
Sent: Friday, July 02, 2004 7:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Aster
Hello All,
Just looking some comments from gurus about this proprietary systems and
phones:
Inter-Tel Eclipse2
Model name: IP PhonePlus
I did not find anything useful or reasonable about their products on
their website or even in Internet except sales.
--
Thanks and regards,
Vasyl Rublyov
goes with CoS and
QoS as well.
Can any one suggest which is better for this choice?
--
Thanks and regards,
Vasyl Rublyov
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are allow, sometimes it helps to use traceroute to
see the path a packet is taking.
Michael Rowley MD
FP
--
Thanks and regards,
Vasyl Rublyov
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov
Sent: Friday, May 28, 2004 4:45 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with PPP internet T1
Patrick,
I seen the problem with HDLC on kernels 2.4.20, but this is explainable.
Did you try compile zaptel
netif_rx(skb);
}
#else
hdlc_netif_rx(ms-hdlcnetdev-netdev, skb);
#endif
#endif
It should be hdlc_type_trans to handle frames.
Vasyl Rublyov wrote:
Michael,
I am going to check the difference between code for 2.4.19/20 and for
latest 2.4.26, to understand
Joseph,
Good point, thanks.
I tried but did not find anything really good, so just opened new page
at http://www.voip-info.org/wiki-Asterisk+Data+Configuration
Feel free to update :-)
Joseph wrote:
There were a number of messages about using
a digium card as a data t1 (router) kind of thing.
Is
Michael,
But really I see the code is commented, not the question - what was the
reason, will it work properly and as designed or breaks something else?
This is from the latest driver code
#if 0
skb-protocol = hdlc_type_trans(skb,
ms-hdlcnetdev-netdev.netdev);
#else
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--
Thanks and regards,
Vasyl Rublyov
IonIdea, Inc.
3913, Old Lee Highway, Suite 33B
Fairfax, VA 22030
Tel
gateway ip netmask isp
subnet mask -arp
Now we can ping our serial ip, but can't ping the isp gateway ip. ifconfig
shows us transmitting packets, but we don't receive any. Any help would be
greatly appreciated.
Thanks,
Patrick
--
Thanks and regards,
Vasyl Rublyov
IonIdea, Inc.
3913, Old
that are in the new kernels we will want to
use... Especially as 2.6 becomes the defacto standard. What then?
I wish I could help you with the driver part. I would love to have
this fixed Just don't have the knowledge, or the time...
M
On Monday, May 24, 2004, at 08:19 PM, Vasyl Rublyov
everything This sucks
M
On Sunday, May 23, 2004, at 12:40 PM, Vasyl Rublyov wrote:
Thank you Michael,
I used that sethdlc which is in latest zaptel, sethdlc --version does
not work, but "sethdlc hdlc0 --version" works
sethdlc --version
Steve,
Do you know buy any chance some info for HW Echo Cancellers which can be
applied on the top of T1 PRI, E1 PRI... as well as to regular analog
(FXO/FXS) trunks/lines?
Thanks,
Vasyl
Steve Creel wrote:
I have a channelized T1 coming in from our telco, terminated onto a TE405.
There are
PROTECTED]]On Behalf Of Vasyl Rublyov
Sent: Friday, May 28, 2004 12:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with PPP internet T1
What is your kernel version?
Patrick J. Conroy wrote:
Hello all,
We have a TE405P set up with span 1 running to a channel bank
/asterisk-users
--
Thanks and regards,
Vasyl Rublyov
IonIdea, Inc.
3913, Old Lee Highway, Suite 33B
Fairfax, VA 22030
Tel: (703) 691-0400
Mob: (703) 395-0238
Fax: (703) 691-0401
www.ionidea.com
A CMM Level III and ISO 9001 Company
-
This e-mail (including any attachments
So I see noone is intersting to fix this and willing to live with old
kernels. Am I right?
Vasyl Rublyov wrote:
All,
Just now I tried Linux kernel 2.4.19, with old sethdlc utility...
everything works, so the problem seems in the zaptel driver or HDLC
implementation of Linux.
I really can't
Brain,
I believe that space on the disk become very cheap, as well as computer
memory, so it does not make any sense to have decoding mpegs in the
memory, consuming cpu cycles for this let save a little for codecs
and echo cancellation :-)
Anyway - thanks for good points - I will
ous to see you get this working, as I need to work on this
next. But I have to get the phone system up and running first.
Michael.
On Saturday, May 22, 2004, at 10:09 PM, Vasyl Rublyov wrote:
Thank you, Michael
I tried to switch to FR mode... but it did not help. I
Christian,
Opss.. than where is the problem? Is it in kernel or is in zaptel driver?
Could you please let me know when has been broken and if anyone is
working on the fix for it?
Thank you.
Christian Hoffmeyer wrote:
- Original Message -
From: Vasyl Rublyov [EMAIL PROTECTED]
To: [EMAIL
and try to fix it,
only wish to confirm that kernel = 2.4.20 works for other T1/E1 cards
with Cisco HDLC protocol.
Vasyl Rublyov wrote:
Christian,
Opss.. than where is the problem? Is it in kernel or is in zaptel driver?
Could you please let me know when has been broken and if anyone is
working
server-list Easy-Servers
# cp 1 ip negotiate-lan no
# cp 1 ip netbios proxy enable no
# cp 1 ip rip receive both
# cp 1 ip rip transmit no
# cp 1 ip multicast-fwd yes
# cp 1 interface-group primary
# ;Netopia 4622
# name ""
# preferences changes immediate yes
# preferences console default menu
# preferences date format mm/dd/yy
# preferences output format verbose
# preferences output mask bits
# preferences time format 24-hour
#
#
=
--
Thanks and regards,
Vasyl Rublyov
this and decode?
Does anyone tried to implement AON?
Thank you,
Vasyl Rublyov
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Just would like to add, of course if it is going to help:
I am using Linux 2.4.26 on Linux, compiled from sources and latest zaptel sources.
We have T1 Internet from Verizon.
... any help appreciated.
= Original message ===
From: Vasyl Rublyov [EMAIL PROTECTED
,
Vasyl Rublyov
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Don,
Just mark another number 703 395 0238... we might help
Feel free to call
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Regards,
Vasyl Rublyov
IonIdea, Inc.
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Feuer
Sent: Saturday, March 13, 2004 10:05 PM
To: [EMAIL
Hi Art,
I am actually interesting more in legal site for this, and most of for
India.
Vasyl
What PBX systems do you have in the US and Ukrain?
There are a couple of ways I believe you could do this.
a) set the PRI port on the Definity as an EM Tie Line, then have * just
perform the VOIP
Hello All,
Does anyone know if it possible to crossconnect PSTN and VoIP system in
India?
I am getting input from local people in India that is not possible due
Laws and found on Internet what is allowed since April 2002.
--
Regards,
Vasyl
smime.p7s
Description: S/MIME Cryptographic
=no
transfer=no
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
jitterbuffers=50
channel = 1-23
--
Regards,
Vasyl Rublyov
[EMAIL PROTECTED]
smime.p7s
-it/archive/2002/03/msg00032.html
Regards,
Vasyl
Hello,
From: Vasyl Rublyov [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Crossconnect VoIP and PSTN in India. Is it
allowed?
Date: Sun, 07 Mar 2004 11:22:53 -0500
Hello All,
Does anyone
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