? (i.e pins 2,5 to pin 3,4 on one port and pins 3,4
to pins 3,4 on the other port?)
i.e. the TDM400P can't span two channels on one port, correct?
Thanks,
-Paul.
___
Correct on all counts.
Good luck with Ameritech ( SBC )
You'll need it!
John Novack
Google finds:
http://www.zultys.com/index.jsp
Linc Fessenden wrote:
Hi everyone!
I just got a zultys zip 2 today with no manuals. Can anyone tell me
how to get in and config the=is thing please? I know there has to be
some *super secret code* to enable dhcp on it somehow and then a login
Rich Adamson wrote:
Assuming you are using the TDM card, there is no code in asterisk to detect
whether a pstn line is connected/disconnected, nor does it listen for dialtone
before dialing.
And for some reason this isn't considered a SEVERE defect?
If the battery on the line
have if it can be done via a T1 port,
or if you have to go in and out via analog.
IF you don't have the Toshiba KSU, then the answer is NO!
John Novack
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Asterisk
.
yours,
matthew
Any clue as to when GOIAX will be back into more or less normal
operation, allowing other than toll free numbers to be dialed?
In the limited time I was able to use goiax, I found it to work well,
and hope it can get back on track
John Novack
be
there with the TDM400, or perhaps the TDM400 and the X100 cards..
I believe some have changed the format of the Vmail , either to or away
from wav, and gotten better results.
Again, search the list archives for answers.
Best of luck
John Novack
Thanks for your answers and patience,
Mike
is a little
slow, you may get misdials
Remember you need to restart after making the change.
To be really safe, reboot.
John Novack
What is correct : pulse=yes or pulsedial=yes ?
However none are working !
I probably miss something.
I need your help.
Is there any french users with pulses
of backlightimg and too small isn't going to be fixed by a firmware
change!
John Novack
Juan Jose Comellas wrote:
I have just bought several Sipura SPA-841 SIP phones, and after some testing I
have found out that the volume received by other parties when calling using
the handset is very low. I've
for some time to come, and needs
to be addressed.
Inserting w into the dial string also seems to not work in HEAD, but
does in 1.0.9, so misdialing persists.
Immediate=no or yes is for incoming calls.
John Novack
i already done (immediate=no) and still it opens the zap trunk even when theres
alarm when there is no battery on the line. Priority + 101 could play an
error message in that case.
John Novack
-- Original Message --
From: Rich Adamson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk
output, there is no way to delay the blind
dialing. Any number of w's are ignored, so misdialing is probable if
for any reason the Co isn't ready.
John Novack
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${EXTEN})
Again, ONLY when dialing DTMF. There is no way to delay dialing pulse,
which, oh by the way, is still used in a number of places around the world!
What is the correct amount of delay?
I have read 1 second, 1/2 second, now 1/4 second. Which is correct?
John Novack
I have changed the zapata and zaptel files from fxs_ks to fxs_gs and
fxs_ls
thinking something is differnet in GA. However nothing seems to affect
the calling out.
Ground start supposedly isn't supported in the X100P and TDM400 FXO modules.
John Novack
if someone knows how to let that be known. I sure don't
or, dig through the w code and copy the process into the dialpulse code.
That is not likely to happen, at least by me!
First the problem needs to be defined,
John Novack
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are sent
out the Zap group to the EM switch
BTW - "w" in the dial string doesn't work here
John Novack
-- Original Message ------
From: John Novack [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Mon, 29 Aug 2005 09:06:29 -0400
b
and there
was a problem, or they never turned it on and tested it.
One problem would be is the whole world doesn't use precise 350/440
dialtone.
It is curious that really cheap modems have been able to do this now for
years.
John Novack
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time, but that is a discussion that can wait for another time.
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, and is
bored by Wayne, Celene et al, Lincoln NE has better attractions. At
least Lincoln has a Telephone Museum!
John Novack
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http
that would require me modifying emailing the guy and either going with cvs or modifying the gentoo ebuild I'm using (bleh).
If that will give you dialtone detection, that's probably what you will
have to do then!
What is NANP?
North American Numbering Plan
JN
Cheer
luck
John Novack
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Then you go to IVR, VM or ??
John Novack
For users within asterisk domain who actually use Dial command it does
not matter and I know that I can have full control over them
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if it doesn't is try another Motherboard
Have you mentioned your version of Asterisk and your flavor of OS?
John Novack
somesh s wrote:
I tried modprobe wctdm also same result...
Regards,
Somesh S. Shanbhag
--- Fabrizio Mazzoni [EMAIL PROTECTED] wrote:
Try using modprobe wctdm
Regards
in
the US, valid only for party lines that rang from T or R to ground
Certainly not a problem in the Gulf Coast these days either.
John Novack
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Sipura's
work great.
John Novack
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( such as
when you pick up another phone ) so it has no way of knowing other than
the ring signal disappearing.
I suspect there is some way to shorten that interval. Given the
different ring cadences around the world it probably should not be too
short though.
John Novack
Works with 1.2 bets!
Voice is MUCH better than festival!
John Novack
trixter http://www.0xdecafbad.com wrote:
I wrote a very very simple shell script and an even simplier macro to
use the IBM TTS engine within asterisk for prompts. While its free you
are limited on the number of requests
to have
been answered and billing would start.
John Novack
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all suffer at the hands of our
politicians
John Novack
Michael D Schelin wrote:
Why is it you have to put down the United States? Sprint is a U.S.
Company. Vonage a U.S. Company. Digium a U.S. Company. What in the
heck does it have to do with you? Even if Vonage is tied up in court
the rest
Eric ManxPower Wieling wrote:
Andrew Kohlsmith wrote:
On Friday 14 October 2005 20:50, Eric ManxPower Wieling wrote:
Oddly enough, I believe it's mentioned in UPGRADE.txt.
Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt.
Sorry, it's in
not detecting
dial tone can result in many mis (pulse) dialed calls to the PSTN
So YES, quite a few still use, and will continue to use pulse dial
[EMAIL PROTECTED] was not used as the configuration for our more or less
special needs proved too cumbersome.
Good luck.
John Novack
Cyrille DERORY wrote
I get a similar warning with 1.2b1
Anyone have a clue as to what this means??
John Novack
asterisk183 wrote:
Why Asterisk show this message:
Nov 30 17:05:17 WARNING[1351]: chan_sip.c:9600
handle_response_register: Got 200 OK on REGISTER that isn't a register
Thanks
Is the 941 available yet?
What improvements have they made.
Backlit display?
The 841 with the late firmware is a fairly nice phone, but the display
might as well not be there. And whoever decided button placement, size
and color should have been fired.
JMO
John Novack
Cory Andrews wrote
, it's just the ZIP2s
that aren't sending the pass.
Have you found a way to get the MWI to work on these phones??
Personally I don't think much of the ZIP2 , overpriced, lack of any
speakerphone or display, no PoE and no wall mounting ability.
John Novack
Thanks in advance happy new
Use one of the clone X100 cards, available on eBay for 10 bucks or so
Not all PCI Voice modems work
Only ones with a certain chipset
John Novack
Matt Murphy wrote:
Hi all, I'm a freshfaced asterisk n00b, and I've got a dumb question.
(tm)
I'm experimenting with an asterisk at home
or the EU?
John Novack
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to search on
cellsocket
John Novack
Brian McEntire wrote:
Is anyone familiar with cell phone switches that allow routing cell
phone calls through in-home wiring? One example of these devices is
the Phone Labs Dock-N-Talk. It says it keeps your cell charged when
you are home and connects your cell
I gather from the information that this is not usable in the US market,.
It is NOT a world band GSM .
John Novack
Sam Tam wrote:
We have ran out of stock in our office in UK. All GSM Gateway are now being
send from HK therefore the shipping will be more expensive than usual.
-Original
with Cisco) and it will probably cost more...
The same can be said for Asterisk, of course.
VOIP is still a baby. Low cost and open source is no substitute for
quality and reliability. Until both improve, it will remain a small
portion of the market.
John Novack
answered once dialing is complete, ,
which complicates billing, and any hope of another action on a real no
answer.
The writers never made any provisions for lines that provide any sort of
answer supervision
John Novack
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( or whatever
number you like ) has gone unanswered, but no way to know for sure.
John Novack
Mouss Greg wrote:
Thanks John for your quick answer,
You're right, i'm trying to put a strong billing system in place,
after many months using ASTCC, i'm now integrating a2billing which
seems to be stronger
[EMAIL PROTECTED] wrote:
1 FXO per PSTN, so you would need 16 FXO ports. That would be accomplished
by 4 TDM100P with 4 FXO modules on each.
Undrhil
Which would be a neat trick in one machine!
HA!
Much better the Sangoma A200
John Novack
--- Asterisk Users
Mailing List - Non
Sure, but if one needs that many, much better off to use the Sangoma
A200 No MB problems and up to 24 channels.
John Novack
Fabio wrote:
Hi Tj,
yes, you can run two TDM400s (or more) on the same cpu, and the channels are
1 to 4, and 5 to 8. (also, you can set one or more groups
and some good protection on the
PSTN line.
John Novack
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Wouldn't it be nice if this fellow fixed his time and date?
John Novack
Khaled Chehab wrote:
I added a context and assigned on it an international number to be
dialed from the asterisk directory *411 as shown below
But the call doest succeed (declined) how can I let directory use my
The destination number has call forwarding, and forwarded it to a number
that may not be correct??
John Novack
Innocent Evil wrote:
NVM,
This is happening for a specific destination numbre only.
I am considerting that destination number have a wired problem.
Thanks,
--
You don't have any
to stay in the path.
Details please
John Novack
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Also many so-called legacy hybrid PBX switches have had this for many
a year
Hard to compete when well used features that have been around for 20
years are lacking
John Novack
Rushowr wrote:
The reason he might want it is because it's a feature offered by many POTS and
Mobile Telcos. I know
That is Camp on
I believe what the OP wanted to do was to dial a number EXTERNAL to
Asterisk until the call went through
Auto Redial and Camp On are NOT the same
Telephony 101
John Novack
Philipp von Klitzing wrote:
Look here:
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
Petty bureaucrats make up their own rules the world over!
Whatever you do, it will be wrong
John Novack
Kevin Kiely wrote:
Be careful here... Our local PSAP is handled by the fire department. I had
one of our guy's make a test call and we were told that this test must be
coordinated
Also, unfortunately, Asterisk does NOT listen for dialtone before
dialing, so these problems will continue until someone sees fir to fix it.
As an aside, for those who pulse dial, rather than DTMF, the w will
not work, as it only works in DTMF
John Novack
Rusty Dekema wrote:
It's normal
to make effective use of their creation
How many have struggled through the years with uncommented or poorly
commented code when the original creator is off to greener pastures?
JMO
John Novack
Moises Silva wrote:
Douglas. Please take this as a constructive comment. I have followed
your
IAX based outboard device with FXO and/or FXS that overcomes the many
shortcomings of the IAXy
Should also support pulse dial, server address expressed in other than
an IP, and all the good features of the SPa 2000 and 3000
JMO
John Novack
Dovid Bender wrote:
I have spoken to some one who
John Novack
Jonathan Borden wrote:
I was wondering which of these cards would be better for a 1-2 line
SOHO. I would like room to grow as well as I am concerned with voice
quality and life expectancy of the product. Any input into which one
and why would be greatly appreciated.
Thanks
to cut the
fixed disk drive bays out of the way to make it fit.
John Novack wrote:
The A200 is a far better card. More forgiving of Motherboards, MUCH
more expandable, slightly lower cost. Only real drawback is modules
are in pairs, so if you want 4 FXO, you need to buy 4. It expands to
24 ports
are a breath of fresh air. I can only hope your outlook and
attitude are contagious in this community and the programming community
in general
John Novack
If I just made something that I needed...and then decided to release
it to
the public incase anyone else needed it...I wouldn't
Be aware that Cellsocket is a dead end.
WHP wireless is gone
When these are gone, they are gone
They do work, however.
GSM versions do not forward CLID
GSM requires a # to end the dial string
They do work however
John Novack
Don Fanning wrote:
How about a Cell Socket? Just plug it into your
I was unclear regarding the CLID
I meant CLID on an INCOMING call.
Cellsocket passes cellsocket, not what the phone received.
John Novack
Steve Kennedy wrote:
On Sun, Aug 20, 2006 at 04:06:06PM -0400, John Novack wrote:
Be aware that Cellsocket is a dead end.
WHP wireless is gone
When
Someone smart with C needs to rework wctdm
John Novack
Chan Kwang Mien wrote:
Hi,
Does anyone know how I could configure the make/break ratio of pulse
dialing in Asterisk ?
regards,
Kwang Mien
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daylight
savings?
Thanks
--
Zeeshan A Zakaria
And with the change in the US next year, adding three weeks in the
Spring and one week in the Fall, where will ALL those smart DST devices be??
John Novack
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towards the Sangoma A200
John Novack
Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank over
2-3 FXO cards ?
Thanks.
Dovid
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of history.
JMO
John Novack
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) wireless number.
I have to wonder why you care.
John Novack
Dovid B wrote:
Does anyone know if there is a way to get a DB or any other means to
see if I can see if a call is coming from a cell phone or not. If I am
able to see if it is cellular or not is there any way to see aprox.
what area
, and as Cox gets
slower, you will have to go back and insert yet another.
No one seems to be interested or skilled enough to fix this.
It's much more fun to add new wiz bang features than to fix some
fundamental design flaws.
John Novack
Dovid B wrote:
snip
How hard would it be to have asterisk detect a dial tone ?
I really can't say. I am not a C programmer, so I wouldn't even know
where to start.
Given that cheap dial up modems have, for the past ??20?? years, have
been able to do just that, I would think it should have
is the wise choice, in my opinion.
John Novack
These are rather compelling reasons, but I'd still like some feedback
from the list prior to making a purchase.
Both cards offer hardware echo cancellation for an additional cost.
Do the benefits of hardware EC justify the expense in this scenario
abridged extension numbers.
Has this been fixed? Or is it considered a feature?
John Novack
Julio Arruda wrote:
Try to search for the PrivacyManager application.
It does 'check' if the CallerID is present, if not, it will play an
announcement to ask the person to 'type' their phone number
for models of a
decent hybrid system, and build on that.
JMO
John Novack
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!
HTH
James
Set in indications.conf, though I believe that changes the Dialtone of
an FXS port on the TDM400 as well.
John Novack
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that
there is only one dialtone specified.
That wouldn't be a bad feature to have, but isn't sexy enough for anyone
to want to work it out.
John Novack
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off the GUI and have at it.
John Novack
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Curious
Additional comment: The Tor3 driver needs to be loaded FIRST, if it is
to co-exist with a TDM400
John Novack
Jefferson Carvalho wrote:
Hello All,
Someone have the govarion phone number?
I am sending e-mails to Mr. Ben and he never
repply me. I buyed a card last week and after
traffic would stop. At least 2 different 4 port devices.
The 8 port version has been fine.
John Novack
Dovid B wrote:
Hi list,
Can anyone who has successfully ran asterisk on a home router please
give me the modell number as well as how they did it ?
Thanks.
Dovid
Not listed as one that is.
JN
bails wrote:
Just out of interest are they openwrt compatible?
Bails
John Novack wrote:
I have been using the Linksys BEFRS81 Version 2 8 port router for
some time now, using IAX to and from other Asterisk boxes, and before
that the 4 port version
Carla Schroder wrote:
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run
into some gotcha down the road where there is some missing file
Derek Whitten wrote:
John Novack wrote:
snip
That sounds like a microsoft way of doing things.. install 25X more crap than
you will ever use. What ever happened to planning and RTFM?
I guess it all depends on what the objective is.
One can sit around and RTFM and play with oneself
installs a bunch of unnecessary
services that you will want to turn off using chkconfig service_name
off.
David
It MIGHT be useful for SOMEONE to specify what those unnecessary
services are
John Novack
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buses.
Sangoma works with MANY more motherboards.
John Novack
jason wrote:
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a
X100P FXO card. These can be had on ebay for 11 bucks, but I
understand that even that pushes the bank some days. You don't need
the card, you only
it takes is some geek 80 miles away to screw with the DNS, and
everything is out for 10-20 minutes.
The modern computer industry still isn't up to the reliability standards
of the telcos.
John Novack
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then recompile Zaptel?
It is normally set to a fairly low voltage
John Novack
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answer try another motherboard choose Sangoma every time
None of that foolishness
Sangoma gives REAL support, not to mention a 5 year Warranty
John Novack
Rudolf Ladyzhenskii wrote:
Thanks for suggestion.
Just tried that Was surprised with download size of only 72k. Anyway,
command work, but I
Are you in the US?
If so, such blocking is not legal, and you should file a complaint with
the FCC
John Novack
Mail list wrote:
Hello my isp has blocked outgoing and incoming connection for port
5060 . I have ssh access to server so i want to send all traffic
from port 5091 to port 5060
Google is your friend!!
http://www.eweek.com/article2/0,1895,1773983,00.asp
http://www.eweek.com/article2/0,1895,1773832,00.asp
http://www.eweek.com/article2/0,1895,1772661,00.asp
Let us hope SS isn't a communications lawyer
The FCC DOES have jurisdiction
John Novack
Steve Sobol wrote
lately. That driver wasn't listed in the
recent past.
Contact David at Sangoma for excellent support
John Novack
Todd- Asterisk wrote:
Hi everyone
I just ordered a Sangoma A20001 with 2FXO ports - Does anyone have
suggested reading pointers for what I'll need to do to get it
working? I've
REALLY don't want to
be in the way of the army of bureaucrats.
Don't talk yourself into inaction. If in fact you have any evedence at
all, then register it.
John Novack
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assignments now mostly left in the laps of bureaucrats, the
NANP will soon be in the same mess as the rest of the world.
John Novack
Doug Crompton wrote:
Wow what a mess! I can imagine how much easier it would be if the world
adopted a country/area/exchange scheme like in the US with known length
different opinions, but have not found a statement of the correct order?
Download and untar zaptel, libpri,asterisk and asterisk add-ons, then
compile and install zaptel, libpri and asterisk THEN asterisk add-ons,
or should it be zaptel, libpri, asterisk add-ons, then finish with Asterisk?
John
information to determine free on network
calling
Since wireline to wireless LNP, the NPA assignments are no longer locked
to a specific carrier.
John Novack
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some time and weather along with a short
commercial were sponsored by banks and such, often recorded on an
Audichron system.
John Novack
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and supports it.
This a pure and simple issue of corporate greed on the part of SBC/att.
John Novack
Dog is my co-pilot
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wants is a simulation of the awful Nortel system
tone on hold used when no other source is available, and the customer
doesn't think silence is better.
John Novack
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cause a macro related to ENUM to fail. Smarter
heads than I have so far been unable to uncover the cause.
Even for those who don't place our business in the hands of the whims of
Asterisk, there are few reasons to make the change simply to have the
latest and greatest?
John Novack
--
Dog is my
Steve Edwards wrote:
On Sat, 15 Dec 2007, Johansson Olle E wrote:
I wonder if there are any major obstacles for upgrading.
How about the change from a bad command line interface to a really bad
command line interface?
I mean, Seriously? (in a Grey's Anatomy kind of way...)
.
John Novack
Cheers,
Daniel Cole (CCNA)
Technical Support
Ph: 1800 424 683
Fax: 03 5221 7659
e: [EMAIL PROTECTED]
w: hugonet.com.au
---
The information transmitted is the property of HugoNet
that is corrected will it make fixing the CDR defect easier?
In other systems with SMDR/SMDI/CDR the report contains information and
timing for the total call, as mostly no one cares who answered the call
and then transferred, it is total time that is important for billing.
John Novack
--
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the
flash card?
Well it IS an appliance, after all.
And that IS the American way! Throw it out when done with it.
We do it with appliances, wives and our pets!
John Novack
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Andrew Joakimsen wrote:
ATT or Verizon. I think those are the only ILECs left, right?
Don't forget the company formerly known as US Worst Now Quest
John Novack
On Dec 31, 2007 9:26 AM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Senad,
Mind if I ask who that provider is?
Thanks
.
John Novack
John covici wrote:
Hi. I have a client who wants some way that his analog phones can
call out even after the power is out and the UPS has died -- some way
that a phone can connect directly to an fxo or some such when power is
gone. Any hardware around which can do this? I have
to reorder. Asterisk shows it trying to go to
a non existent context, not specified in sip.conf
Any ideas?
John Novack
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That is not the name the phone requests
When uping my 7960, the empty file did the trick
I so far am unable to go beyond 7.1 however, as Asterisk rejects
anything I dial with 7.3
Anyone have SIMPLE sample config files?
John Novack
Anciso, Roy wrote:
Try naming the empty file:
SEP0019E7D16CD6
and
erroneously called RJ11
8 Position plugs with 8 contacts used for many applications, including
Ethernet
10 position plugs - seldom if ever seen.
For more detail:
http://en.wikipedia.org/wiki/Modular_connector
John Novack
Glenn Cobb wrote:
Here is the info on the Cisco console cable I use
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