Hello All,
I have been using the following phones with excellent success inside my
lan:
Cisco 7960
Polycom IP600
I have also been comfortable with the SPA-3000.
I recently got a SPA-841 and the quality is aweful. Even in stock
setup, it's like when I speak into the handset I sound like I am
Hello All,
Has anyone tried to tie an LCD display into asterisk? This could be a
neat function. Mostly to monitor the number of concurrent calls, and
cpu usage...
I have some extra lcd's but they arent connected at this point, would
like to use them for something...
I could bounty $30 for
I did that once on a cheap linejack card. Took a week to get the smell
out of the office, and the bright orange from inside the server was
quite interesting :) Only took 1 second to start a small flame going,
but fortunately I cought it quick.
I wonder if the zaptel cards have any kind of
I was having problems and your tip helped, my handset showed a polarity
reversal... Now we'll see how well it works...
Thanks,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, June 08, 2005 2:27 PM
To:
I had an odd issue too. Seems that the only fix was, unplug poly,
restart asterisk, plug in poly. The registrations would drop. This is
behind a non-nat vpn.
When I did a provision as opposed to register, it seemed to work better.
Asterisk would get abunch of not authorized messages.
Greg
In an ip600, I just setup 6 different numbers on the phone, 1 per line.
All was fine.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Ford
Sent: Monday, May 30, 2005 3:33 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
It is likely that they took a 7970, and superimposed video on it. While
this is uncertain, tv programs very often superimpose video ontop of
computer screens, etc. Even if there was a video phone from cisco which
they used, they would be required to superimpose. If you've noticed,
video cameras
I agree that voipsupply is great! Now can we end this thread? So they
had phone problems, so what. As for the license issues, that is not
their problem. If cisco is a pain about licenses it's cisco's issue.
Sometimes I have secretaries that just fail to show up. I think we can
live with it
OMG! Has anyone tried to visit www.ken-ton.com?
It's a laugh! Explains his whole
email.
Heh...
Greg
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl J.
VesterlingSent: Wednesday, May 25, 2005 1:29 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionCc:
Title: Asterisk-Users Digest, Vol 10, Issue 174
Hello Adam,
I am busy all day until after 6pm EST. If you want
you can call me then at 1-914-591-2211.
Regards,
Greg
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
CollardSent: Monday, May 23, 2005 12:54 AMTo:
Why not do this. For that extension, in your dialplan, do a busy status
detect. If busy, send to a Queue for that line only. Should do the
job, and when the phone is available it will start ringing.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Title: Asterisk-Users Digest, Vol 10, Issue 174
Hello Adam,
Did you ever get this solved?
Greg
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
CollardSent: Monday, May 23, 2005 12:54 AMTo:
asterisk-users@lists.digium.comSubject: Cisco 7940g Firmware load
problems
I
I have a pingtel hardphone, and while it's nice, I suspect that asterisk
will kick butt over pingtel's ambitions. An example is, sipfoundry does
not have the support or ease of use as asterisk. I suspect they are in
it for the enterprise business model, which is good, but all the nice
features
Hello Rod, I'll try it, thanks.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Wednesday, May 18, 2005 1:01 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO
Make sure you have disabled
Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On May 17, 2005, at 19:00, Gregory Wiktor - ADCom Corp. wrote:
On a recompile of the kernel I now get a 99.98 average.
Static is gone, although
, 2005, at 19:00, Gregory Wiktor - ADCom Corp. wrote:
On a recompile of the kernel I now get a 99.98 average.
Static is gone, although quality so far seems not quite there yet.
I am also experiencing an odd local echo. I can hear a slight echo
locally, but the other end sounds fine, and the other
Hello All,
Figured I'd ask this. I have an asterisk running with some X100P
clones. The system runs fine.
My question is, the lines are tied into a panasonic dbs analog
extension. When you call through, all goes well except that the call
does not know it is connected. This causes problems if
16, 2005 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO
On 5/16/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote:
Hello All,
I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy
static.
Even
Hello All,
I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy
static.
Even with the pots line disconnected, if I do a dial I still get static.
This way I know it's not the line, but rather something on the card.
I tried alternate pci slots.
This card has a power connector,
Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Gregory Wiktor - ADCom Corp. wrote:
Hello All,
I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy
static.
Even with the pots line disconnected, if I do a dial I
-Original Message-
From: Gregory Wiktor - ADCom Corp.
Sent: Monday, May 16, 2005 5:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO
Hello,
Yes in the bios I set unused irq's. Zttest is sporadic.
This is a non
PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Monday, May 16, 2005 5:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO
Hello,
Yes in the bios I set unused irq's. Zttest is sporadic.
This is a non HT cpu
As a test, I had a 7960 and a polycom, and made direct
calls between the two. I transferred the begeezez out of them back and
forth, sending to an incoming queue, etc. Asterisk performed very
well. Even over a vpn cable connection
(polycomroutercablemodeminternetcablemodem
Hello Guys,
I suspect you misunderstood. What I am lokinng for is a hardphone that works
well. Asterisk itself is running fine... The issue is for example, if in
europe, behind an uncontrollable nat, are there good phones that can handle
this without requiring the configuration of the
Yes I saw you mentioned the sipura phone was good for this... I will
probably pick one up and check it out...
Thanks... Next time I should read through the list decending rather
than ascending...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have had success with 5, but if you have 6 you can have a
hard time transferring with no lines left, unless you use
features.
The correct usage is more like 11 though, since you can use
call waiting...
Must you use cisco? I like both cisco and polycom,
but I beleive both polycom snom
I am a user of Teliax and voipjet. I find voipjet to be very reliable
and good for outgoing, very low lag, etc. Teliax is good too, but I am
finding high lag rates, to the point where there is a half-second delay.
I ended up just ordering a pots line for incoming (since I am going to
be doing
Hello All,
I am looking for a sip phone that is capable of automatic nat. The
Cisco ata186 for example works fine for natting with iconnecthere, but
as for asterisk, both my 7960 and polycom ip600 require you to set the
nat ip on the tftp.
Does anyone know a good phone (or ata) that can do this
I had been trying to get mwi working on a pingtel phone for some time,
with no success...
The solution was simple. When I made my voicemail.conf, I added the
boxes to the end of the file. The problem was, at the end was in a
different context, so mwi would not light. The solution, all I had to
Hello All,
Does anyone know how to reduce the incoming ringtime on the polycoms?
What I mean is, When I have an incoming call, my 7960 and pingtel ring
immediately, but the polycom seems to delay 2 seconds before ringing...
Any ideas?
Greg
___
Me too...
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Harrison
Sent: Tuesday, April 12, 2005 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: polycom phones
On Apr 11, 2005 11:49 PM,
Erase your caller directory, happened to me because the default ring on
directory was 1 (silent)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Sent: Thursday, April 07, 2005 12:24 AM
To: Asterisk Users Mailing List - Non-Commercial
Why not go with Multitech? They are expensive, but great units.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
George
Sent: Friday, May 06, 2005 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Either its suspending or APM, or watchdog... Check your hardware...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, May 06, 2005 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
@lists.digium.com
Subject: RE: [Asterisk-Users] Verizon ISDN
Greg, the Diva Server cards are around $900 for a single BRI and $2500
for a Quad. The Adran unit can be had for less than $400 used.
Brian
On Wed, 2005-04-13 at 16:30, Gregory Wiktor - ADCom Corp. wrote:
Brian,
I am looking into the diva
Systems
888.288.5470
www.multitech.com
-Original Message-
From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 04, 2005 4:01 PM
To: Multi-Tech Sales
Subject: RE: MVP410-ST
Hello Sales,
You mention you do not support 5ESS, do you mean your tech support dept
or does
I setup my 7960 with line 1 as main, and 2 as a queue line. So if the
line is busy, asterisk queues the call and it will continue to ring on
line 2. Call waiting works too, but not as well as queueing...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I actually setup 6 registrations as separate lines. This allows me
dialout selection, like line 5 for teliax, line 6 for voipjet, etc.
I suspect you need different logons or all of your lines would ring at
once.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
So you have the receptionists voice right, then she goes for coffee and
someone else picks up, that would be odd... :)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Pratt
Sent: Monday, May 02, 2005 3:00 PM
To: Pedro; Asterisk Users Mailing
Hello Sean,
I thought the Polycom's had some kind of BLF Feature don't they?
I am thinking of getting two of them, so it would be nice to know,
otherwise I would get 2 more 7960's. (which are great phones)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I know I saw something about not using GSM codecs when on cell phones, could
this be the case?
The 2 second delay, well unfortunately all cell's have about a .5 second delay
on their own, so that may be what you are hearing. You just need to learn how
to talk like you are on an international
I just made an extension 390 that calls my cell, so people can hold,
then send to 390 and hangup.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 7:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Hello Tomek,
I also got a diva pci 2.02 card, but although the kernel sees the
incoming calls, asterisk refuses to answer. Did you have this issue at
all?
The kernel seems to be denying the call...
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
: Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server)
workwithasterisk!
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
I also got a diva pci 2.02 card, but although the kernel sees the
incoming calls, asterisk refuses to answer. Did you have this issue
at all?
The kernel seems
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Thursday, April 28, 2005 3:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver)
workwithasterisk!
Gregory Wiktor - ADCom Corp. wrote
I use Opexld.com for my 800 service, then send the number to a did.
More reliable this way in case a provider has problems.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Thursday, April 28, 2005 9:29 AM
To:
Hello Tomek,
I suspect I may have located my problem. The Hisax driver supports
NI-1, but I believe my CO is sending signalling in 5ESS, which does not
seem to be supported by the hisax driver
Looks like it's back to the drawing board...
Greg
-Original Message-
From: [EMAIL
for a Quad. The Adran unit can be had for less than $400 used.
Brian
On Wed, 2005-04-13 at 16:30, Gregory Wiktor - ADCom Corp. wrote:
Brian,
I am looking into the diva isdn cards for around 200 at this point. I
had the semiactive but it did not work, though it is S/T so I may be
able to use
I am happy with teliax...
Keep in mind though, I am using 800 from opex, who can do an instant
change to another number. That goes to a local did. This way, if
something happens and I am very busy, I can just call opex and they can
switch it.
Its like 2.9 cents per minute or so, on top of the
Maybe you just got a broken one?
If you need high end, the multitechs are good products, with excellent
support.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
Sent: Monday, April 25, 2005 2:28 PM
To: Asterisk Users Mailing List -
I just got a cisco 7960, a bit tough to get going at first
but it's a great phone. Supports OHVA, and the dialplan is very nice in
that you can have autocompletion based on your plan. for example, if I dial 300,
the phone completes, whereas if i dial a 1+ number there is a
timeout.
For
I just wrote a simple cgi to have a form generate the number, then the
cgi creates a call file and bingo. Web call.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Elkins
Sent: Friday, April 22, 2005 8:21 AM
To: asterisk-users@lists.digium.com
Hello All,
I was wondering how everyone got along with cisco 7960's. I just picked
one up and I am having problems locating an image. I called cisco, but
they will not sell to end users... Does anyone know a place where it
can be purchased in the US?
It has stock firmware, and the skinny seems
How about a linksys wrt54g with sveasoft firmware? Has some shaping and
many other nice features...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Max Clark
Sent: Friday, April 22, 2005 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Questions about a 7960 and images
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Gregory Wiktor - ADCom Corp. wrote:
Hello All,
I was wondering how everyone got along with cisco 7960's. I just
First, you may want to consider that you do not have enough zap
channels. Can you tell us something about your system? How many lines
do you have, and are you bridging incoming calls to an extension or
flashing them through a pbx?
-Original Message-
From: [EMAIL PROTECTED]
Hello All,
I have a setup I would like to try as such:
Call comes in.
Rings to a sip extension, if busy, goes to menu
If not busy, follows other procedures.
My question is, I want asterisk to ring 3 extensions simultaneously, but
if the first one is busy, to put the call into a queue.
This is
Seems odd, though I would suspect the boards.
Have you tried higher end boards, like compaq proliant servers?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Monday, April 18, 2005 11:25 AM
To: 'Andrew Latham'; 'Asterisk Users Mailing
PROTECTED] On Behalf Of Walt Reed
Sent: Monday, April 18, 2005 8:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?
On Sun, Apr 17, 2005 at 01:50:56PM -0700, snacktime said:
On 4/17/05, Gregory Wiktor - ADCom Corp
Hello All,
I have been trying a did company for a few days. I find the service
decent, but sound quality only moderate.
Rather than spending 35 or so for monthly with did, I am considering an
isdn bri at this location.
How much more stable and reliable is bri or pri versus a voip did
service?
I have to agree that voipjet is a good service. If only they had did's
it would be even better, but I like the fact that outgoing cid works
well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Greenberg
Sent: Saturday, April 16, 2005 6:35 PM
To:
Hello All,
I was wondering if anyone has a solution to the following:
I have a dialplan to route to an extension, but if the extension is busy
I want the user to get a message, and hold. I want them to be able to
press 1 to leave a message if they do not wish to hold.
If the extension rings but
the wiki for campon. It may help point you in the right
direction.
- Original Message -
From: Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 16, 2005 3:43 AM
Subject
Lucky you, my spa-3000 likes to dial 911. So far the local cops have
been nice about it though. (my mobile number ends in 9110)
I have been having trouble getting quality and tone transmission right.
Seems to be a delay, but there is less than 5ms ping time. G729 is 'ok'
and ulaw seems
Hello,
Does anyone know of a script that can take a voicemail, and deliver it
to a mobile phone or pbx vm system?
For example, I have a panasonic dbs voicemail, which has vmwi and
telephone based vm navigation. I want to accept vm's on asterisk, then
later forward the vm to the pbx, by for
Of course, you can have the telco put on rj-11 jacks and just run them
to the tdm. Plus consider an analog fxs port for fax, etc.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Bruce Lynes
Sent: Thursday, April 14, 2005 2:49 PM
To:
I recall seeing though, that they may not show up as lit buttons,
meaning you may not neccesarily be able to see status.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Thursday, April 14, 2005 2:57 PM
To: Asterisk Users Mailing
Simon,
Looks like its not seeing your card. Which capi modem are you using?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Morris
Sent: Wednesday, April 13, 2005 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
, Gregory Wiktor - ADCom Corp. wrote:
Brian,
I bought a diva pro pci 3.0 on ebay. Turned out since it's a
semiactive card it will not work, but got a terminator too. So I am
still looking for a Diva PCI or Diva server pci without spending a
bundle. Just got the line in today, Verizon took
is and if you don't
mind where you bought the card,
thanks,
Brian
On Tue, 2005-03-29 at 22:38, Gregory Wiktor - ADCom Corp. wrote:
Next week I have an isdn coming in, if it happens, odd area. I am
going to try an EICON Diva Pro PCI, I'll make a post when I get it
running.
Though without
Brian,
If you spoke to eicon, any idea if they have some pci cards laying
around? They seem impossible to find on resellers.
Server is great probably, but the price is too steep. Its almost easier
to use zaptel analogs tied into a normal isdn modem, except then there
are no features.
Regards,
Next week I have an isdn coming in, if it happens, odd area. I am going
to try an EICON Diva Pro PCI, I'll make a post when I get it running.
Though without Centrex at this point.
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian G
Best solution, use a regular pbx with asterisk doing
cross-transfers and using cid for identification. For example, I am happy with
Panasonic DBS for a front-end pbx with asterisk behind it.
Regards,
Greg
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
DevitoSent:
Neel,
I managed to do this with a simple CGI script. All it does is take the
phone number from a form, and create a call file on the asterisk
machine. Then when my side answers, the call completes to the other
side.
Look into call files and you can see what I mean...
Greg
-Original
Hello Marc,
Could you post the install details for the Diva PCI? I have a Diva Pro
PCI 3.0 and I am looking to get it working... I am running Debian
Testing. So far it appears there are a few things floating around but I
am not certain which direction to go. I know I need to load the firmware
Hello,
Does anyone know which EICON card is best with Asterisk in the USA?
There are a few, but I was wondering should I get an eicon diva 2.0, or
and eicon diva pro?
Thanks,
Greg
___
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