I just went through a Dahdi rebuild, and I seem to recall a message that
all modules will be loaded until you set up the dahdi configuration files.
regards
Scott
On 7/9/2010 11:41 AM, Gilles wrote:
Hello
To use Dahdi + Asterisk with a PCI card with a single FXO port, I
just...
1.
Hello-
For maintenance purposes, if possible I'd like to use the same iax.conf
file in several different asterisk systems. However, on one of the
systems only, I would like to include an IAX register command to
another external system.
Within iax.conf or other configuration files (other
On 7/7/2010 11:25 AM, Danny Nicholas wrote:
--
Rather than trying to determine what system you are on, just make the
included file be empty on all except the desired server.
OK, thanks. I thought I might have to do it that way, which is slightly
less desirable, as it makes the systems
On 7/7/2010 11:52 AM, Kevin P. Fleming wrote:
On 07/07/2010 01:46 PM, Scott Stingel wrote:
On 7/7/2010 11:25 AM, Danny Nicholas wrote:
--
Rather than trying to determine what system you are on, just make the
included file be empty on all except the desired server.
OK
On 6/30/2010 3:56 PM, Alex Villacís Lasso wrote:
whenever an ISDN port is in RED alarm (unsynchronized), we get a stream of
warnings in /var/log/asterisk/full that look like this:
[Jun 30 17:38:41] WARNING[9637] chan_dahdi.c: No D-channels available!
Using Primary channel 78 as D-channel
On 6/22/2010 4:26 AM, A J Stiles wrote:
Is anybody else using the following combination:
* a TE410P card (wct4xxp driver)
* a BT ISDN connection
* DAHDI 2.3.0.1
* Asterisk 1.6.2.9
I'm trying to configure a new box to replace a legacy system (same hardware;
some old version of Asterisk
On 6/22/2010 2:03 AM, Tzafrir Cohen wrote:
On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote:
Hello-
I have a system with one D410P and one B200P (both OpenVox). All is
well with the D410P, inbound and outbound, and I can initiate calls on
the B200P BRI span
Thanks to the OpenVox engineer for picking this up:
I had bri_cpe for my signaling type, should be bri_cpe_ptmp. The
BRI circuit on the B200P works fine now in both directions.
-Scott
On 6/22/2010 7:58 AM, Scott Stingel wrote:
On 6/22/2010 2:03 AM, Tzafrir Cohen wrote:
On Mon, Jun 21
On 6/22/2010 9:44 AM, A J Stiles wrote:
On Tuesday 22 Jun 2010, Scott Stingel wrote:
Hi-
I've been going through the same upgrade process recently, and had the
same error (shown in your other message). I had forgotten that the
equipment I was plugged in to was CPE, so I had to change my
On 6/22/2010 2:15 PM, A J Stiles wrote:
[Jun 22 21:34:18] WARNING[5651]: chan_dahdi.c:4160 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
AJ-
On my system (D410P) the above message appears when EITHER:
(a) A span is configured in
Hello-
I have a system with one D410P and one B200P (both OpenVox). All is
well with the D410P, inbound and outbound, and I can initiate calls on
the B200P BRI span, but there may be something wrong with my inbound
BRI setup: there is no indication of an inbound call when I dial in to
it
On 6/17/2010 6:47 AM, Gareth Blades wrote:
Scott Stingel wrote:
On 6/17/2010 2:12 AM, Gareth Blades wrote:
Scott Stingel wrote:
Hello-
After configuring DAHDI and starting asterisk, I get the following
message continuously on the Asterisk console:
WARNING[2057]: chan_dahdi.c:4158
Hello-
After configuring DAHDI and starting asterisk, I get the following
message continuously on the Asterisk console:
WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels
available! Using Primary channel 16 as D-channel anyway!
My card is a D410P configured for E1, only the
I think there's an Asterisk-Biz mail list for this purpose..
-Scott
On 5/29/2010 6:12 PM, Robert Augustyn wrote:
Can anybody provide DIDs for Chatham, ON?
Usage based preferred, but flat-rate is not an issue.
Contact off list.
Thanks for your time,
Sincerely,
Robert Augustyn
--
Hello all-
My client has purchased these two OpenVox cards and I'm configuring a
system with Asterisk 1.6. In the past I have used bristuff and libpri
with older versions of Asterisk, but now I would like to upgrade to
Asterisk 1.6. Question, should I be using mISDN or libpri for these
On 5/27/2010 2:33 PM, Philipp von Klitzing wrote:
Hi!
Question, should I be using mISDN or libpri for these cards when they
are in the same system, or does DAHDI now support both cards under
asterisk 1.6 reliably?
I cannot answer that question, but do stay away from mISDN if
, which is described as a Q-Sig version of 2BCT. Anyone have
an idea whether this will work with the Rockwell ACD?
Thanks in advance,
Scott Stingel
www.evtmedia.com
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Thanks Julian - will update and see if it works.
regards,
Scott
Julian Lyndon-Smith wrote:
We are using 1.4 trunk with sms - it got fixed recently (about 4-5 weeks
ago I think).
Julian
Scott Stingel wrote:
Hi all-
In 2004, I set up a sms texting process for a UK customer, using
application I
would think, not a crossover, but I mention it anyway.
Best regards,
Scott Stingel
evtmedia.com
Paul Hales wrote:
I am setting up an E1 crossover system for a customer, with a Siemens
Hipath Officecom 150 system.
And it's not working - I get a red alarm from the outside world
, and if so, can you please shed some light on this?
Many thanks
Scott Stingel
www.evtmedia.com
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in the asterisk bounties section on
the Wiki, and someone added a note saying that this might be possible in
a 5ESS configuration.
Has anyone actually used TBCT under asterisk?
Thanks
Scott Stingel
Emerging Voice Technology, Inc.
www.evtmedia.com
telephone jacks wherever you want (called inside wiring), but usually
at quite a high hourly rate.
Regards
Scott Stingel
www.evtmedia.com
Manjit Riat wrote:
Thanks but I am pretty sure they won't do it. So there has to be a better
way.
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent
LeeLee-
Try configuring all 4 spans first and then the single channel (125)
above that - works for me.
Modprobe in the same order, then ztcfg.
Regards
Scott Stingel
www.evtmedia.com
Lee Lee wrote:
Hi all
I newly added a X101P into my asterisk that already have a TE410P
running 2 E1s namely
office?
Do you have another line you can switch to and try the same card?
Does the Red alarm occur at the moment the call is disconnected, or
afterward?
Regards
Scott Stingel
www.evtmedia.com
Yusuf Iqbal wrote:
Previously I have posted the same mail but no one answered me...Sorry
for resending
to me
that they are at a point of growth where a significant investment is
required to allow growth to the next level.
Regards
Scott Stingel
www.evtmedia.com
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Good question. I've had good luck with the Digium TE405P recently in a
multiple T1 install.
Don't let the discussion scare you too much!
-Scott Stingel
cmould wrote:
Where is this discussion going. I am about to do an installation that
will require t1 interfaces. I am new to the telephone
Bravo - nice writeup Matt!
It concisely captures both the pros and cons.
Seems that we really do have (or are close to having) a second source
now - and all asterisk users will benefit in my opinion!
Cheers
Scott Stingel
www.evtmedia.com
mattf wrote:
My Sangoma Experience in Asterisk
of alphabetical order as the list
requests. My edit was promptly re-edited, so I gave up!
Anyway, just one of the risks of an open Wiki!
regards
Scott Stingel
Emerging Voice Technology
www.evtmedia.com
Sean Kennedy wrote:
Hi folks,
I recently registered with the wiki site to fix a few things I've
THAT ztcfg
commented out, Junghanns issue
#sleep 3
startast
sleep 3
GOOD LUCK
Scott Stingel
www.evtmedia.com
-
Victor Alvarez wrote:
Hello,
I have a machine with two cards installed, one digium that gives e1
connectivity and one quadBri for the ISDN line
would be in order?
It's just bad PR for the company when these things happen.
Please keep us up to date on what happens.
regards
Scott Stingel
www.evtmedia.com
[EMAIL PROTECTED] wrote:
Hey gang,
Just pulled out our brand new $1,500 TE405P and put it into a Dell Poweredge
6450. Nothing. Card
spans, but heavy
transcoding might cut that number down somewhat. Test before you commit!
Hope this gets you started.
Regards
Scott Stingel
www.evtmedia.com
Wai Wu wrote:
Hi all,
My first post here. We will be implementing an * solution with Digium
Pri cards. I have a few questions. First, what
of performance and reliability.
-Scott Stingel
www.evtmedia.com
Andrew Kohlsmith wrote:
I respectfully disagree. Sangoma's voice capabilities are no less and no more
mature than Digium's voice capabilities.
I use cards from both Sangoma and Digium. Both seem to work well but (and it
does pain me
(Thread moved from -dev)
Hi Ma-
You probably can do this with a 3GHz+ processor, but I don't have experience
with such a busy voicemail server - perhaps someone on here will...
Regards,
Scott Stingel
www.evtmedia.com
Just to confirm that you also powered down and up?
I've no experience with the TE110, but this is a known problem with the
TE405 and TE410. They apparently can get locked up, and only a power
cycle will clear it.
Regards
Scott Stingel
www.evtmedia.com
Alfredo Sola wrote:
Hi,
I have
From looking at the description, it seems that the Sangoma card (at
least the quad version) *may* have a more robust hardware buffering
mechanism than the TE4xxP series. If so, this might help solve some of
the load-related issues that my customers have experienced in very large
systems.
an
IVR call to a live operator without holding up channels in the IVR.
Regards,
Scott Stingel
Emerging Voice Technology, Inc.
www.evtmedia.com
Alistair Cunningham wrote:
Eric,
E1 is a physical layer protocol, like ethernet. It defines a 2Mbps
pipe, which can be used for data, or can be split
to send calls from one to the other. By the
way, to initiate outbound calls use the .call file facility (see the Wiki)
Good luck!
Scott Stingel
President
Emerging Voice Technology, Inc.
Palo Alto California and London England
www.evtmedia.com
Johan Bilien wrote:
I guess I need some special equipment
you may not be aware of the asterisk-biz mailing list, which is probably
more appropriate for a discussion like this.
you'll find many VoIP termination vendors hang out there too.
Regards,
Scott Stingel
Mohit Muthanna wrote:
Have you used them before?
Do they provide commercial grade service
!
regards,
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
Sören Malchow wrote:
Hi all,
i am stuck with the configuration of asterisk
- modules are loaded ( zaptel and wct4xxp )
- i have zaptel.conf configure, output of ztcfg -vv
--- snip --
/rapid:~# ztcfg -vv/
/Zaptel
in. If
you're used to Outlook, thought, you may miss the nice integration with
your calendar, but you certainly won't miss paying for it!
-Scott Stingel
www.evtmedia.com
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http
-48m volt power is often used in telco central office environments,
where the C.O. provides a huge amount of battery-backed up power to the
switches and to power the local loops in the event of an AC power failure.
Regards
Scott Stingel
Emerging Voice Technology, Inc.
www.evtmedia.com
You can read all about it, and find out where to download at:
http://www.voip-info.org/tiki-index.php?page=Asterisk
Yes, it supports both SIP and H.323
Cheers
Scott Stingel
sai latha wrote:
Hello,
Happy New Year
where u r downloaded the asterisk server please
tell me.Iam searching the asterisk
Andrew-
Thanks for posting your update and troubleshooting checklist.Most
people on the forum don't take the time to re-post when a problem has
been resolved - but that's the thing that helps people the most!.
regards
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
on the other end
Pin 2 -- Pin 5
Pin 4 -- Pin 1
Pin 5 -- Pin 2
You should get green's on both the connected channels if your zaptel and
zapata configurations are ok, and if you've run both modprobe and ztcfg
as documented.
Good luck
Scott Stingel
President
EVT, Inc.
www.evtmedia.com
Sid wrote:
Hi list
: NOT on this mailing list):scott at
evtmedia.com,
ie: do not reply to this, just send me a new email and please put:
asterisk consulting or something in the subject line so I can see it
among the spam!
Thanks
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
for perfection and just use a very
fast processor when I want to handle 120 channels with no errors.
Best regards
Scott Stingel
President
Emerging Voice Technology, Inc.
www.evtmedia.com
---
[EMAIL PROTECTED] wrote:
Dear All,
we have installed a TE410P card
!
Regards
Scott Stingel
_www.evtmedia.com
_
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I use SetCallerID() and it displays the number just fine on my
GrandStream phone.
Regards,
Scott Stingel
www.evtmedia.com
David Ishmael wrote:
I'm confused, should I be using SetCallerID(${CALLERIDNUM}) or
SetCIDNum(${CALLERIDNUM})? Also, I don't think it matters but I'm trying to
forward
correctly). This allows you to see if the problem is outside or inside
the system.
Good luck!
Scott Stingel
www.evtmedia.com
Ciro La Ferrara wrote:
Hi,
I am new with asterisk. I am setting a Wildcard TE405P. E1s in Italy
come in on a pair of RG-59 coax cables with BNC connectors. So I need
This is not exactly leading-edge, but my customers are running 15 asterisk
busy asterisk servers, all on Fedora Core 1, with kernel updates. Pretty
solid
Don't know about Core 3.
Cheers
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California
Hi-
You probably would be better off trying to sell this board on Ebay or
something, although I notice even there they are not selling for very much.
This board was retired a few years ago by Dialogic.
The effort of writing a driver for this board in the asterisk environment
would be huge - why
line
and I'll do it for cost. (but you'd have to pay the sales tax of 8.5% since
I'm in California like they are!)
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL
I would say that the Auto Attendant function, used to allow people to select
who they want to talk to, is a subset of IVR, which more generically allows
the person calling to retrieve information or to control the operation of a
computer or telephone switch.
Scott M. Stingel
President,
Emerging
Peter-
I haven't retried this lately, but it worked fine in the past when I did.
Be sure that you follow the instructions in the README exactly, especially
the notes about which versions of pwlib and openH323 versions work with the
current version of OH323
Regards
Scott M. Stingel
President,
Matthew:
Not sure if this is the problem, but I usually compile in this order,
different from yours:
zaptel
libpri
asterisk
asterisk-addons
regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original
.
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
- Original Message -
From: Geraldo Fco. do Espírito Santo Jr. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 12, 2004 9:21 AM
Subject
Brian-
I was quoted (verbally) something on the order of $60 per month for a single
BRI by SBC in San Francisco about 60 days ago. I thought that was high..
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
experience. When checking with NTL ask if your circuit is ISDN 110 or Q.931e
or ETSI; these are the three names that are commonly used by UK carriers for
full spec. EuroISDN PRI.
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
Ning-
I think it goes like this:
TE410P - 3.3v PCI only
TE405P - 5v PCI only
E100P and T100P - 5v PCI only
Better double check with Digium though - to be sure
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
Sounds like the asterisk system is not sourcing enough line current on the
FXS port, and the answering machine may think the line has hung up. This
is what the answering machine is supposed to do, for example, when someone
picks up another phone on the same line.
Regards
Scott
Scott M.
Hi Andrew-
I know you've likely described this before, but what again are the
differences between the FC1 and RH9 implementations? I've always built them
the same way.
Also, what are the other devel and web RPM's that you've posted in that
directory?
Best regards
Scott Stingel
Scott M
If you have a solidly re-produceable bug, suggest that you go to
http://bugs.digium.com/login_page.php
Sign up, and post the bug.
Regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL
Yes, Fedora works fine. Debian too. (I've used both)
...and others have successfully used other flavours... See the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk
Regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
Regis-
I believe the cards are pretty identical, except for the support of the 3.3v
PCI bus on the TE410P, and the 5v PCI on the TE405P.
I've run extensive load tests on both cards in a large IVR environment, with
one processor generating load to the other, and found them perform about the
same.
, or really dual
processors for that matter - unless you're going to start doing transcoding
or other processor-loading tasks.
Please see my notes on the Wiki on this topic:
http://www.voip-info.org/wiki-Asterisk+dimensioning
Regards,
Scott Stingel
Scott M. Stingel
President,
Emerging Voice
Sorry, just dual Xeon's, not P4's!
Scott M. Stingel
Hi-
I've run extensive load testing with both single and dual P4's and
Xeon's (all at least 2.8GHz), and I've got 6 installed IVR systems of
this size in various configurations.
Hmm, I was under the impression that it was impossible to run
There is an asterisk-biz list for this type of post.
Asterisk-users is the non-commercial forum.
Thanks!
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto
That's strange - I'm using wait(10) as the first command for an extension
(followed by an answer) in my dialplan and it works fine.
But on this system, I have a CVS from June.
Maybe you've discovered a recently-introduced bug?
regards
Scott M. Stingel
President,
Emerging Voice Technology,
factors,
including the number of channels, how many will be transcoding ,and other
things.
Here's the link: http://www.voip-info.org/tiki-index.php?page=Asterisk
Also, many readers on here would prefer that you post your messages in plain
text, not HTML.
Regards
Scott Stingel
Scott M. Stingel
Yet another forum seems quite unnecessary to me, but I added a post there
referring people to the Wiki.
Scott Stingel
www.evtmedia.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Keating
Sent: Friday, September 17, 2004 10:06 AM
zaptel.conf looks good - you may require a ,crc4 at the end of the span
line, depending on your provider. I have loadzone=us in mine as well.
Change appropriately for yours.
zapata.conf also looks good. I would also add the following, before the
channel declaration:
immediate=no
Steve-
That's an interesting/amusing story! The only thing I would worry about is
using the Zapateller SIT tone as the first thing whenever there's no caller
ID. In many places (like here in California), a good percentage of people
have caller ID blocking on outbound calls from their home
Could you please post this on asterisk-biz instead?
Each email here is sent to over 8000 people, surely this is not a suitable
place to conduct the poll you're looking for.
Regards
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
It's normal, in fact I use it to be sure that everything's ok, since I think
it will not occur unless we have no alarms on the spans!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
Hi Adam-
I'm wondering if the TE405P might be a better choice, since it's 5 volt PCI
and may allow you to consider a wider selection of motherboards. Sounds
like you may not need the latest and fastest motherboards, which often use
the 64bit 3.3v slots as you've probably found.
Regards
Scott
Scott.
I have used servers from advansor, one with a 2 Xeon cpus, 2 nics, hw
raid and a te405p card, and another with 1 P4 cpu and 1 t100p, both working
veri well.
The only bad thing is that advansor site has an Altigen add ;-p
Scott Stingel wrote:
Hi-
I have an upcoming order
Answering machine detection is usually accomplished by analysing the timing
of the voice energy in the initial answer period. People usually answer by
saying: Hello, Frank Giwerski, Pencil sharpening department, or
something fairly short, whereas answering messages are usually longer.
So, I
Hi Julian-
I was using a BT BRI line, with caller ID option enabled. Also, I had to
send 1470 before the call because my customer had blocked his outgoing
number on this line.
So I'm certain that it works on BRI. BT says in their SIN document
(Supplier's Information Note), number 413, that
?
Thanks
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup
Sent: Tuesday, August 31, 2004 7:14 AM
To: [EMAIL
Claus:
One difference is that I'm using the slower ATA disk, not the SCSI.
Is the noise rhythmic (periodic) or constant? If periodic, what is the time
between noise bursts?
Do you hear the noise throughout a call, or just occasionally?
Regards
Scott Stingel
Scott M. Stingel
President
to change in the next couple months when BT will accept SMS's
from all carriers.
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
You should be able to hear the audio - a sound card is not involved.
Try inserting an answer command in the dialplan before you try to play
something. Like
Answer
Wait (if you want)
Playback
Hangup
Should work (using the proper dialplan commands)
Regards
Scott Stingel
Scott M. Stingel
You'll find the following web site to have a huge amount of information (too
much really!)
http://www.voip-info.org/tiki-index.php?page=Asterisk
Regards
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
with asterisk?
Thanks in advance
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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.
BUT, I can't get the Dell's fast enough for this customer, so now I'm
looking at the HP Proliant DL-320.
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
Flynn:
As far as max capacity, you might want to check out:
http://www.voip-info.org/wiki-Asterisk+dimensioning
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
than the rest of the machines in the
server room combined.
Craig
- Original Message -
From: Scott Stingel [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Wednesday, August 25, 2004 2:56 AM
Subject: RE: [Asterisk-Users] Dell PowerEdge
Except for the PCI bus voltages, there does not appear to be any difference.
I've load-tested both extensively and they perform about the same.
I did have an issue with a TE410P getting stuck, ie not responding after a
re-boot (but not a power down), but that seems to have resolved itself when
://www.voip-info.org/tiki-index.php?page=Asterisk
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stig
test too. I understand that
there are software VoIP load generator programs that can do just this.
If you are using T1's or E1's, I hope that my script will be useful in some
way, as I found that there is a call setup/teardown load too when lots of
calls are handled.
Regards
Scott Stingel
Meet-me is a generic term, came about some time in the 70's. I think the
first use was in hospitals, where people would be paged on a beeper. The
operator would park the call, page the doctor, and he/she would then call a
number that would meet an incoming call.
Cheers
Scott Stingel
Scott M
-93
channel = 94-108,110-124
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz
Sent
-108,110-124
Regards
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz
Sent: Thursday, August 12, 2004 2
and it didn't make any difference. :o(
Gary
- Original Message -
From: Scott Stingel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 11, 2004 5:39 PM
Subject: RE: [Asterisk-Users] zaphfc problems...
Have you tried changing pridialplan and prilocaldialplan to unknown
-
From: Scott Stingel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 12, 2004 11:03 AM
Subject: RE: [Asterisk-Users] zaphfc problems...
Gary-
Try turning on BRI debugging (not intense), and post it. (just the part
with errors that you get during the connection). Maybe you'll
Hi-
Anyone using the Junghann's quad BRI card and the Digium E100P in the same
system?
I'm having a configuration problem where I can configure the cards one at a
time (with the appropriate drivers loaded) in a system, but when I try them
both together, neither will work. They both work fine
-v wct1xxp
ztcfg -vv** REMOVED THIS
Normally, when I have a system with just an E1, I do a modprobe first and
then a ztcfg, which gives me a reassuring list of channels. I guess the
redundant ztcfg doesn't matter in this case.
Thanks for your help
Scott Stingel
Scott M. Stingel
I'm using vanilla bri-stuff-0.1.0-RC2k from www.junghanns.net and
everything looks right. There are no errors or warnings during startup.
It seems to work correctly (well SIP and IAX2 anyway) until I try to
dial out over the ISDN line.
I get the following:
*CLI -- Executing
It's difficult to give you help when you provide so little information.
Please post your zaptel.conf and zapata.conf at least, and say what your 4
spans are connected to.
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
From:
contributing to the overall load.
Someone should try a multiprocessor solution with 4 E1's and some
transcoding and see where the limit really is. Should be easy enough if you
have 2 boxes available - running 4 E1's on one to generate calls to 4 E1's
on the other.
regards
Scott Stingel
Scott M
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