Yeahit's better to be a KC...we just hang out in the back and watch
everyone else be wrong out loud.
-Jonathan
KC9FQT
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Bowyer
Sent: Friday, September 09, 2005 8:45 AM
To: Asterisk Users Mailing
Hmm a phone system taking in 16 lines for $4,000.There are
companies out there who would sell a phone system like that for $14,000.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: Thursday, September 08, 2005 10:27 PM
Just a poll because I am curious, what kind of call volume
do you see?
Calls/Day
-Jonathan
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http
I'm not using an FTP server :-/ I guess I'll have to setup an FTP server
and have it get it's files from there...it appears that something is
corrupted on the phone.
...
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday
I use them with just the one NIC card. I don't use them as a router so
the phones and my gateway are all on the same network.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, September 06, 2005 4:33 PM
To: Asterisk
The VHF or HF is determined by the radio equipment you have attached, not the
software.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy
Sent: Wednesday, September 07, 2005 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial
You asked how to connect lines, so he answered that question. The answer
is basically the same just change the FXO in the channel bank to FXS.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005 12
to connect many analog lines to
Asterisk?
Jonathan k. Creasy wrote:
You asked how to connect lines, so he answered that question. The
answer is basically the same just change the FXO in the channel bank
to FXS.
Well, actually, I said: If I have more than a hundred analog telephones
(analog lines
. It's empty though. Any
ideas?
-Jonathan
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My polycom phone is now hanging at Running sip.ld.
I modified it's config via the web interface to register with my
asterisk box.
I have tried to restore the default settings wth 468* and it doesn't
seem to work.
Any ideas?
-jonathan
Title: Message
I used
to think that, but now that I have a Polycom IP300 on my desk, the slightly
higher cost is well worth it.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
AmezcuaSent: Tuesday, September 06, 2005 4:22 PMTo:
Title: Message
I
wasn't comparing it to the IP Adaptor. He said "Sipura Sets" and I took that to
mean the sipura handsets like the SPA-841.
-Jonathan
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
AmezcuaSen
, the server sends the
caller to the voicemail box matching the DID. (4 digits in this case for
the extensions originally called which is now going to voicemail)
In any casethe system in question is using an interconnect and I
will be hooking up asterisk via a T1 link.
-Jonathan
Probably already been covered but you do have the power connector on the
board connected right?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
Sent: Friday, September 02, 2005 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial
How does one go about connecting Asterisk to a Meridian PBX to handle
voicemail?
I have a customer who is out of capacity on their voicemail system
(which connects to their meridian via several FXS cards) and I would
like to see if I could use Asterisk to handle their voicemail.
-Jonathan
way to get the info to the phones either.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Wednesday, August 31, 2005 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk
I have had no trouble reaching it.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Friday, August 26, 2005 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voip-info
I have had similar experience with an Intel NIC that had DELL's name on
it vs a 3COM 3C905b.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Logan
Sent: Friday, August 26, 2005 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial
I think he's talking about putting protection from the PSTN lines not
the incoming power.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, August 25, 2005 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial
those anymore.
They were attached to the asterisk server via a 100mb LAN and 1ms was
the highest latency ping would ever show.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Monday, August 22, 2005 9:47 AM
To: Asterisk Users
Is the php script available somewhere?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Gault
Sent: Monday, August 22, 2005 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] All Page ??
I have (sort
w and W allow call recording when passes as options to DIAL, in this
case they are being passed as options to D().
-jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Monday, August 22, 2005 3:51 PM
To: Asterisk Users Mailing List
I'm not having any problems with the SPA-841's at the moment. I have 15
of them in use right now. The other phones we use are the Polycom
IP30X's and they are really nice phones for that price range. I haven't
tried a really expensive phone so I may not know what I'm missing.
-Jonathan
I'm
Do you need a hangup in your dialplan?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard
Leadmon
Sent: Friday, August 19, 2005 4:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Ascend Pipeline
What problem are you trying to solve with this? Just stepping out on a
limb but it sounds like you are trying to swat a fly with an F-16.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Wednesday, August 17, 2005 4:34 AM
to some of our
smaller edge locations or if we want to just use one of these for those
locations with 14 phones.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Graves
Sent: Thursday, August 18, 2005 11:18 AM
To: Asterisk Users Mailing List
No
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 17, 2005 7:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How many TDM22P Card can be used on the same
PC ?
Is it possible to use 24
That's impressive. I have had trouble with more than 2 digium cards in
the same box.
It wasn't even worth messing with, just used a channel bank and a PRI.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Ternero
Sent: Wednesday, August 17
Maybe you didn't intend this for the list?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Thursday, August 18, 2005 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Set voicemail maximum
The power supply could definitely be the problem. You tried a difference
TDM04B right?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Fortin
Sent: Thursday, August 18, 2005 3:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk
Yes, you could do that with Asterisk and Cepstral/Festival.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 6:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] is this possible
I think the foneBRIDGE is too expensive for what it does. IMHO
-jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
Andrews
Sent: Tuesday, August 16, 2005 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk
translations? What about docs and support? What are the
chances the box is really just an mini * server?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy
Sent: Tuesday, August 16, 2005 2:51 PM
To: Asterisk Users Mailing List
You will want to use the D(digitstopluseindtmf) option on your dial cmd.
That is a capital D for the option!
ex.
Dial(SIP/2100,D(1000))
-Jon
Stephen wrote:
Hi All,
Can Asterisk dial extension which resides in the PABX?
(eg. 2000) Sip Phone - Asterisk -- ATA (FXS) --
(CO
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Thursday, August 11, 2005 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk
systemtoreplace an old PBX but using existing phone
Jonathan k
Our vendor told us we can't buy the 841's anymoreanyone else have
this problem or have a vendor that is still selling them?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Ternero
Sent: Wednesday, August 10, 2005 9:44 PM
To: 'Asterisk
how am I to know
it's secure enough not to allow for the creation of a backdoor, rootkit
or anything else?
Therefore, even when I have taken all other security measures I also
lock down a box with a firewall. Usually, iptables, not a separate
firewall device.
-Jonathan
-Original Message
figure out why this one
didn't work. It appears that's all it waqs.
Without the power connecter the card will probe, and even appear to be
working but when the lines ring (coming into the FXO port) it will not
indicate the ring status to asterisk.
-Jonathan
-Original Message
There is pfSense (based on monowall) which I like also. www.pfsense.com
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Sent: Wednesday, August 10, 2005 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
not picky about
where it comes from.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, August 11, 2005 9:44 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Is it mandatory to give power
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
I use the 300 and 301 models. Haven't used them extensively though. The
most common phone for me is the Sipura SPA-841.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
Melanson
Sent: Thursday, August 11, 2005 4:05 PM
To: Asterisk Users
This poll by the company which made amp might be useful to you.
http://www.coalescentsystems.ca/index.php?option=com_polltask=resultsi
d=4
Distribution for your Asterisk/AMP system(s)?
Red Hat Enterprise, White Box, CentOS
48 41.4%
Debian
19 16.4%
Novell/SUSE
The 301 (we only have one) is working, the other phones (there are only
3) are 300's and they are working as well.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun
Bolling
Sent: Thursday, August 11, 2005 4:47 PM
To: Asterisk Users Mailing
The only phones I have much experience with are the sipura spa-841's,
the netweb 301/302 phones (which I really don't like) and the polycom
300/301's. It applies to the sipuras and the polycom's for sure.
I can't remember about the netweb, we quit using them sometime last
year.
-Jonathan
but using existing phone
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm fairly certain
an old PBX but using existing phone
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm
Check out pfSense.
www.pfSense.com
It has SIPProXD on it. This software also has a huge list of truly
awesome features.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bastian
Schern
Sent: Thursday, August 11, 2005 9:41 AM
To: asterisk
Jonathan k. Creasy [EMAIL PROTECTED] writes:
even when I have taken all other security measures I also lock down
a box with a firewall
You still should follow the lists. As long as your computer is
processing data received from the net, it may be vulnerable;)
Generic attacks against the IP stack
Wiley is definitely right. It would be dangerous not to have a firewall
for security reasons.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Wednesday, August 10, 2005 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
connecter the card will probe, and even appear to be
working but when the lines ring (coming into the FXO port) it will not
indicate the ring status to asterisk.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Wednesday, August 10
We had this problem about 8-10 months ago and the end cause was IRQ
scheduling problems with the card.
We put it in a slot with a fixed IRQ and changed something else (sorry,
I don't remember what it was) to fix the IRQ problem and the error
seconds went away.
-Jonathan
-Original Message
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jonathan k. Creasy
Sent: Wednesday, August 10, 2005 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Is it mandatory to give power
supply to TDM400Pcard
Works that way for me. IN SPA-841 for example, both lines are on the
same user/pass and the device registers once but line one rings and if I
answer it then get another call, line two rings.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
. At this point
we are purchasing Sipura 3000s for analog connectivity, mainly because of the
support issue and secondarily due to never being able to get the four port
cards to a stable configuration.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax
200s have been good for us. Reliable good sound
quality, but don't do the presence lights and multi-line stuff the snoms do.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Alexandre Leclerc [EMAIL PROTECTED]:
Hi all,
We
mailbox=sip_line_#. If there is a patch in progress for this, I'd love
to test it out.
Thanks in advance.
-Jonathan
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at the API command interface yet, but I know it's there.
Instead of messing around with the API, how about a timed script which dials
out and tries to connect back in thru Teliax. If the call does not connect
then the script could call/email/alarm you.
Jonathan
Adam Robins wrote:
If anyone out there is running Asterisk with Zaptel and a TDM400P card
on a Dell Poweredge 1850 server, please let me know what OS and kernel
version you are running.
I keep getting errors when modprobing zaptel and am running out of
possibilities, other than motherboard
Andrew Kohlsmith wrote:
BTW are you *really* saving any time by bastardizing your email so much (ur,
u, bcz)... jeez.
I think they teach that crap in school these days ... kids and their sms
cell phones..
Jonathan
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it proprietary without any
valid reason :-)
http://www.voip-info.org/tiki-index.php?page=Asterisk%20protocols
I think that should be corrected!
Happy now? :)
Jonathan / denon
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and hints on exactly how to do this.
Regards
Jonathan
--
Jonathan Gill [EMAIL PROTECTED]
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how to get sound to work with ztdummy and not have to recompile
zaptel every time.
Jonathan Berger
082 574 5064
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-HEAD-02/17/05-11:17:10, on a linux box with GNU bash,
version 2.05b.0(1)
Any ideas as to what would be causing this behaviour are greatly
appreciated!
Jonathan
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The full path name fixed the problem with script execution.
Thanks for all the help!
Jonathan
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: March 9, 2005 9:16 AM
Subject: Re: [Asterisk-Users] Asterisk System() call error
On Wed
Asterisk with a simple agi routine could do this easily.
Jonathan
- Original Message -
From: Eric Balsa [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: March 7, 2005 1:02 PM
Subject: [Asterisk-Users] DTMF to Email
I need some suggestions (not necessarily using Asterisk
To paraphrase:
Ignore them and they will go away.
- Original Message -
From: David Brodbeck [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: March 4, 2005 11:04 AM
Subject: RE: [OT] - [Asterisk-Users] Why should I
'Segmentation Faults'
whenever I ran them.
I gave up on Festival (unable to resolve the segmentation
faults in a reasonable number of hours of effort), but if you manage to get it
running letme know how!
Jonathan
- Original Message -
From:
Wiley
Siler
To: Asterisk Users
How about getting rid of the drives (hard, floppy, cd, dvd) and using
RamDisk technology instead. You can boot from flash memory. This will
reduce heat and increase reliability.
Jonathan
- Original Message -
From: Vledder, Hans [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
What you want is SetGlobalVar, which sets a variable that is available to
any channel.
hth
Jonathan
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: February 24, 2005 10:31 AM
Subject
the TimeOut code did not execute?
Any and all ideas, comments, suggestions appreciated!
Jonathan
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).
hth
Jonathan
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: February 23, 2005 3:50 PM
Subject: RE: [Asterisk-Users] IVR stats
Not a bad choice.. Ive seen software like XT
Hi All,
I am having trouble getting speex to work on asterisk. I downloaded
1.0.4 from speex.org, download libogg from vorbis. ./configure, make and
make install for both and then recompile my asterisk 1.0.5, yet I am getting
error when trying to load codec_speex.so. Below is the error from
In the vain of asterisk in new-zealand...
Anyone know of a reliable source of digium gear in singapore? Also
where to pick up IP phones, anyone any clues?
Ta
Jonathan
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Hi Altus
What sort of price are you able to get? Im only looking for prob 2
(cheap) ip phones right now, maybe more later if all goes well... And as
this is personal stuff, im on a tight budget.
Ta
Jonathan
On Mon, 2005-02-14 at 15:40 +0200, Altus Snyman wrote:
I can get you a good deal
Not sure if this is a helpful answer, but we have looked and haven't come across
anything yet for the US. Curious to see if you get any other responses.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Michael Graves [EMAIL PROTECTED
and leave
the second jack unplugged same result as above, but if
I use only jack 2and leave jac 1 unplugged I see just
the loopback info and nothing about eth0 with the
defined ip and netconfig does nothing.
Any help getting this up and runnign would be greatly
appreciated.
-Jonathan
not seem to be the case. I would be interested in knowing the
configuration and cabling on any successful installations. Thank you.
Jonathan
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On Tue, 2004-12-21 at 01:49 -0800, Jonathan Augenstine wrote:
Has anyone successfully connected a Digium T100P to a Zhone Z-Plex 10 24
S/O? I have been unsuccessful in getting the T1 to sync up. I have
searched the documentation
, but there is some
good info on troubleshooting and error indecations.
I second the post below. Adit 600s are a good choice for Asterisk.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Gregory Junker [EMAIL PROTECTED]:
Say I get
Ciprian Zetea wrote:
Hi Jonathan,
can you be a little more clear ? What is your test configuration? How
do you expect to have voice if you use only one FXO of the card (maybe
you use regular phones too ..)
I have an asterisk box doing VoIP and connecting to the telephone service.
I have local
My TDM400P w/ 4 FXO cards seems to have trouble with onhook/offhook
switching. It dials perfectly, but does not seem to be changing the
onhook/offhook state appropriately. It changes sometimes, but it's not
really reliable. For example:
When I booted the machine, it started as onhook. It
I'm setting up an asterisk server, used as a gateway to regular phone
lines. I've got a TDM400P card with FXO modules, but I'm only using one
to test.
When I make outgoing calls, occassionally it seems like the incoming
audio is switched off. It will work fine for several calls, and then
Kevin P. Fleming wrote:
Race Vanderdecken wrote:
Why can't I convert the DS3 input to SIP Output, no transcoding,
straight G.711, all in one box?
Yes, that is what you would want to do. Probably even better would be
DS-3 to IAX, and try to get trunking support for G.711 working to keep
down
of an existing
keysystem to add new features like voicemail and IVR. It kind of depends on what
your existing cabling looks like as to which makes more sense. The rj11 block is
more expensive, so I wouldn't recommend it unless you really have a good reason
over a 66 block.
--
Jonathan Moore
Director
Below...
Jonathan Moore wrote:
I have a client that uses two queues for a customer support application.
One
queue is for english speaking customers. The other is for spanish.
Try this in queues.conf:
[english]
:
:
member = Agent/1000
member = Agent/1001
member = Agent/1002,5
in the queue.conf file and I thought this
had some impact, but the client still reports problems. Are they handled
alphebetically or some other way?
Anyone know how to resolve this?
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Visit
Can anyone tell me if they have successfully deployed the X100P in India or
any where in Southeast Asia?
Thank you,
Jonathan
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place the call to the party already on the phone, once the party called
from the agi hangs up the process will repeat until phone a hangs up!
I have writen some stuff in php that generates .call files and it so far
seems to be solid.
-Jonathan
Jack Turer wrote:
I am working on a web phone
their products every year (or it least that's what happened with the brand
new gear we've bought from them so far.. i.e. as5200, as5350,
catalyst3500xl). I'd prefer someone else who will provide firmware
fixes/updates without a contract. Say, where's the wiki?
Jonathan Miller
-Original Message
?
Jonathan Miller
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Joe Greco
Sent: Saturday, October 16, 2004 8:58 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Simple phone question
Continuing on/adding my $0.02 to Joe's reply on this thread
The real problem is that the old comdial is dying and cannot support any
more extensions. Not to mention constant static problems on speakerphone.
(even after repunching the all the cables + replacing them).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
The real problem is that the old comdial is dying and cannot support any
more extensions. Not to mention constant static problems on speakerphone.
(even after repunching the all the cables + replacing them). We just really
want a new system that will be expandable for the future.
check out:
http://www.voip-info.org/wiki-Asterisk+phones
At 12:23 AM 10/15/2004 -0700, you wrote:
Hi,
After reading up on the Asterisk, I have a
question:
1. Is there a software phone running on PC as a client that
is compatible with Asterisk?
My reason for asking is that I wonder if I can
, will the different
lines flash when there is an incoming call? If so, how is this
configured/controlled in asterisk? Just trying to figure this stuff out.
Any suggestions or help would be greatly appreciated. Thanks again
asterisk!
Jonathan Miller
___
Asterisk-Users
Not the cheapest ($75-80) but they look interesting.
http://ipphone.eezeephone.com/
Jonathan
At 03:10 PM 9/26/2004 -0300, you wrote:
On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman [EMAIL PROTECTED] wrote:
Hi guys,
I know this isn't strictly about Asterisk, but it is related...
I am looking
Another solution would be to keep the discussions on topic and open up a
separate mailing list for people interested in open discussions.
Jonathan
At 07:17 PM 9/26/2004 +0100, you wrote:
I was somewhat concerned reading Mark's posting earlier today.
Obviously, things are very bad in the US
Asterisk on an SMP system. I cannot find that info now and most of the
reading seems to indicate that SMP stability is good. Does anyone know of
the warning I read?
Thank you.
Jonathan Augenstine
[EMAIL PROTECTED]
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Asterisk-Users mailing list
[EMAIL
the feature.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Noah Miller [EMAIL PROTECTED]:
Hi -
I'm wondering if any has experience with the Uniden UIP-200 phones.
The product info says that the 8 led buttons at the top are all
the
problem by using my configs. So there is at least some awareness at Uniden that
people are using this on *.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Ryan Courtnage [EMAIL PROTECTED]:
Jonathan Moore wrote
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