On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan
<[EMAIL PROTECTED]> wrote:
> Matt Florell wrote:
>> Hello,
>>
>> I guess I am one of the lucky few to have one of these handy
>> screwdrivers and it saved me when my son(aged 2) somehow locked
>> himself in a bedroom and couldn't unlock the door. Th
On Tue, Jun 17, 2008 at 12:07 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> On Mon, Jun 16, 2008 at 11:11:00AM -0400, Steve Totaro wrote:
>> On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
>> > On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote:
>> >> Is t
On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson
<[EMAIL PROTECTED]> wrote:
> But maybe an AVM Fritz! box will work for you too...
Would anyone care to recommend a good quality, stable ATA these days
for just a single cordless phone connected to one SIP provider. Sipura
used to be well thought-of
On Mon, Jun 16, 2008 at 10:43 PM, Jared Smith <[EMAIL PROTECTED]> wrote:
> By "tweaker", I assume you mean the small screwdrivers we often give
Actually, I think I overstepped the true definition of "tweaker". This
is a real screwdriver with anti-geekdom feature. It looks like a pen
when it is stu
On Tue, Jun 17, 2008 at 1:22 AM, Mark Hamilton <[EMAIL PROTECTED]> wrote:
> How come he has it, and he's in Paris! I'm in Toronto, and I don't have it?
> :(
There was an Astricon in Paris. After my presentation, I was awarded
the "get screwed" gift pack!
__
The screwdriver is reversible, it swings both ways, pull out the shank
and stick it in the other way, it becomes a Phillips. I'm tellin ya,
there Digium engineers are good!
(Phillips is a trademark of Phillips)
On Tue, Jun 17, 2008 at 3:28 AM, Richard Lyman <[EMAIL PROTECTED]> wrote:
> Mark Hamil
On Mon, 16 Jun 2008, Eric Fort wrote:
> I'm presently working on provisioning VoIP to a traditional key system. I
> have a single SIP DID inbound that gives me a maximum of 2 concurrent
> channels. I need an ATA that will ring the second station port when the
> first is in use. What devices wil
Bikrish Amatya wrote:
> Hi all
>
> I am using asterisk as pbx for my company. My company has requirement
> that all the incoming and outgoing calls should be recorded for all the
> extensions and should be able to play recorded call on extensions basis,
> that is , say 123 extension has made wha
Thanks all for your responses. I will look into the Asterisk Queues and
VICIDIAL Call Center Suite.
On Mon, Jun 16, 2008 at 11:44 PM, Sherwood McGowan <
[EMAIL PROTECTED]> wrote:
> Syed Nasruddin wrote:
> > Dear Sherwood,
> >
> > I am also using Asterisk Call Center Setup in my office with voice
Hi all
I am using asterisk as pbx for my company. My company has requirement
that all the incoming and outgoing calls should be recorded for all the
extensions and should be able to play recorded call on extensions basis,
that is , say 123 extension has made what call on the particular date
an
On Mon, Jun 16, 2008 at 09:30:12PM -0400, Matt Florell wrote:
> Any chance of more of these being handed out at Astricon this year?
Oh, sure... beat me to the idea. :-)
BTW: did the annotated syllabus notes from the Boot Camp ever finally
get finished? I never got a copy.
Cheers,
-- jra
--
Ja
On Sat, Jun 14, 2008 at 06:34:26PM -0700, John Todd wrote:
> I did receive one objection so far to the fifth track because of the
> fear of missing a good talk in the other four tracks. However, I'm
> not sure how we overcome that problem if we don't add the track, as
> we would have to dump 11
On Mon, Jun 16, 2008 at 11:11:00AM -0400, Steve Totaro wrote:
> On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> > On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote:
> >> Is there a contradiction between them?
> >
> > Naw; Steve's just showin' his ass again
I'm presently working on provisioning VoIP to a traditional key system. I
have a single SIP DID inbound that gives me a maximum of 2 concurrent
channels. I need an ATA that will ring the second station port when the
first is in use. What devices will do this with a single sip registration
with t
Matt Florell wrote:
> Hello,
>
> I guess I am one of the lucky few to have one of these handy
> screwdrivers and it saved me when my son(aged 2) somehow locked
> himself in a bedroom and couldn't unlock the door. The door knob
> needed a very small slotted screwdriver to twist-unlock the door and
>
Syed Nasruddin wrote:
> Dear Sherwood,
>
> I am also using Asterisk Call Center Setup in my office with voice
> recording. The only thing I am unable to setup is web based call
> recording (CDR) access. From your email I think you have configured such
> a thing can you please share with me how can
Syed Nasruddin wrote:
> Dear Sherwood,
>
> I am also using Asterisk Call Center Setup in my office with voice
> recording. The only thing I am unable to setup is web based call
> recording (CDR) access. From your email I think you have configured such
> a thing can you please share with me how can
Dear Sherwood,
I am also using Asterisk Call Center Setup in my office with voice
recording. The only thing I am unable to setup is web based call
recording (CDR) access. From your email I think you have configured such
a thing can you please share with me how can I also setup this solution.
I kn
On Mon, Jun 16, 2008 at 8:24 PM, broadband Voice
<[EMAIL PROTECTED]> wrote:
> Is anyone using Asterisk as a call center. I want to be able to set it up
> for my office line, when calls come in after 7:00pm Est want a recording to
> says the office is closed and have about 5 phones that I want to us
Hello,
I guess I am one of the lucky few to have one of these handy
screwdrivers and it saved me when my son(aged 2) somehow locked
himself in a bedroom and couldn't unlock the door. The door knob
needed a very small slotted screwdriver to twist-unlock the door and
the Digium tweeker(which was als
Mark Hamilton wrote:
> Now you're just trying to get us all jealous, Steve. No good.
> But I'd like that screwdriver!
>
>
I hope JT is taking notes and will get the higher ups to add 'tweakers'
to the digium store.
On a personal note, i still haven't seen my 'sticker'! haha
___
Hello,
We have set up dozens of call centers, some using Asterisk Queues and
the rest using VICIDIAL Call Center Suite. What you want can easily be
accomplished with an average server and Asterisk Queues with not too
much effort using standard Asterisk configuration features. we have
set up a smal
Now you're just trying to get us all jealous, Steve. No good.
But I'd like that screwdriver!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: June 16, 2008 8:41 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discu
broadband Voice wrote:
> Is anyone using Asterisk as a call center. I want to be able to set it
> up for my office line, when calls come in after 7:00pm Est want a
> recording to says the office is closed and have about 5 phones that I
> want to use as an agent. Can anyone share their implementa
I had a laser pointer and power point controller device but the Digium
logo rubbed off after a week I do have a t-shirt though
Thanks,
Steve T
On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists)
<[EMAIL PROTECTED]> wrote:
> On June 16, 2008 07:22:18 pm Mark Hamilton wrote:
>> How c
On June 16, 2008 07:22:18 pm Mark Hamilton wrote:
> How come he has it, and he's in Paris! I'm in Toronto, and I don't have it?
Yeah, me too. I even got a mention in the book, but no screwdriver? :-(
-A.
___
-- Bandwidth and Colocation Provided by htt
Is anyone using Asterisk as a call center. I want to be able to set it up
for my office line, when calls come in after 7:00pm Est want a recording to
says the office is closed and have about 5 phones that I want to use as an
agent. Can anyone share their implementation? Thanks.
I've been playing around in order to find something new and I've found this:
I have created an IVR for test purposes, then I've placed a call from my sip
phone using one of my telco lines to another of my telco lines attached to
the PBX, in this situation I'm using two FXO channels, one for the ou
read the fine prints with them. Sometimes they have a thresh hold. Where you
cannot exceed 1500 minutes per month on the trunk. I use a T1 which is
cheaper if you compare it.
On Fri, Jun 13, 2008 at 4:52 PM, Jonn R Taylor <[EMAIL PROTECTED]>
wrote:
> I use bandwidth.com, works very well. 5 trunks
> How come he has it, and he's in Paris! I'm in Toronto, and I don't have
it?
> :(
I was thinking the same thing, Ottawa here.. :(
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
Mark Hamilton wrote:
> How come he has it, and he's in Paris! I'm in Toronto, and I don't have it?
> :(
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
> Sent: June 16, 2008 4:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Dis
How come he has it, and he's in Paris! I'm in Toronto, and I don't have it?
:(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: June 16, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Benoit Plessis a écrit :
Hi,
I'm having trouble with a TE220p PRI card and (outbond) caller
identification.
Previously with usecallingpres=no everything was Ok, one small
difference between the
BRI (B410p) was that the callerid needed to be stripped from one number.
But then came the need
Hi Steve and the rest of the list,
On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro <
[EMAIL PROTECTED]> wrote:
> Is your Asterisk box dual homed? Firewalled? Any output from the CLI
> with verbose turned on, that might help? Turn on SIP debugging as
> well.
>
> Thanks,
> Steve T
>
>
My Asterisk
I just hafta ask, why does one face down a requirement for 48 FXOs?
Would it not be more practical to have 2 T-1s dropped into the
location?
Michael
On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote:
>Adit 600 48 FXO.
>
>On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku <[EMAIL PROTECTED]> w
Adit 600 48 FXO.
On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku <[EMAIL PROTECTED]> wrote:
> Steve,
>Thanks for the responses. I am talking of 45 POTS
> Thanks
>
> Steve Totaro wrote:
>
> Sorry,
>
> Quantify "High Traffic"
>
> How many POTS lines are we talking about?
>
> Thanks,
> Steve Totar
On Sun, 2008-06-15 at 18:04 +0200, randulo wrote:
> Moving day, everything packed. Including tools! But wait, there in the
> jar with pens and pencils... it looks like. Yes, it's the Digium
> Asterisk tweaker!
>
> THANKS Digium!
By "tweaker", I assume you mean the small screwdrivers we often give
On Mon, 2008-06-16 at 15:01 -0400, Dean Collins wrote:
> thanks moderatorit was a perfectly reasonable email - just
> oversized because of all the urls.
> "No reason given"
>
This is most likely my fault... Just for clarification, any message
over 40k gets moderated. I thought I told the ma
thanks moderatorit was a perfectly reasonable email - just oversized
because of all the urls.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Mon 6/16/2008 1:55 PM
To: Dean Collins
Subject: Request to mailing list asterisk-users rejected
Your r
Steve,
Thanks for the responses. I am talking of 45 POTS
Thanks
Steve Totaro wrote:
Sorry,
Quantify "High Traffic"
How many POTS lines are we talking about?
Thanks,
Steve Totaro
On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
I use Adtran or Adit, I think Rhin
Brian,
I agree with you. Google has all the answers, but not the
experience. The reason I use lists is to get opinions 'experienced'
users. From experience, product manuals say one thing and when the
rubber meets the road, its a different story.
Thanks though for your kind comments
Brian
He is talking about a free Digium Screwdriver
On Mon, Jun 16, 2008 at 11:07 AM, Andrew Kohlsmith (lists)
<[EMAIL PROTECTED]> wrote:
> On June 15, 2008 12:04:01 pm randulo wrote:
>> Moving day, everything packed. Including tools! But wait, there in the
>> jar with pens and pencils... it looks l
On Mon, Jun 16, 2008 at 12:30 PM, Kyle Sexton <[EMAIL PROTECTED]> wrote:
> Having a weird issue with some agents getting stuck busy on my system. Call
> will come into the queue and the agent will hit DND, or be DND when the call
> comes in (DND being the button on eyeBeam softphone, not a star co
On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote:
>> Is there a contradiction between them?
>
> Naw; Steve's just showin' his ass again.
>
> Cheers,
> -- jra
Nah, just showing various tactics, sure some con
On June 15, 2008 12:04:01 pm randulo wrote:
> Moving day, everything packed. Including tools! But wait, there in the
> jar with pens and pencils... it looks like. Yes, it's the Digium
> Asterisk tweaker!
>
> THANKS Digium!
>
> Before you ask, it's 1.0 I think.
?
-A.
_
On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote:
> > Happens in the commercial world all the time; it's a common way to "get
> > cash out of the corporation" -- a business's building is owned by the
> > corporation's owners, and rented to the corporation.
>
> This is actually illegal in some s
On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote:
> Is there a contradiction between them?
Naw; Steve's just showin' his ass again.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think
Hi,
I'm having trouble with a TE220p PRI card and (outbond) caller
identification.
Previously with usecallingpres=no everything was Ok, one small
difference between the
BRI (B410p) was that the callerid needed to be stripped from one number.
But then came the need to make "hidden" calls, and
Just FWIW, I have been doing double NAT with asterisk and all kinds of
SIP phones for years, including BT101, Sipura 941, and Polycom IP500
plus many cheap no name brands, plus many softphones like Zoiper,
X-Lite and Gizmo Project.. However, DMZ has never worked properly for
me with asterisk on any
Gordon Henderson wrote:
> On Mon, 16 Jun 2008, Gary Guthary wrote:
>
>> If I take a device to my office (i.e. the Sipura) and connect it. - It is
>> configured to 'talk' to my home's 'public IP'. - This thing doesn't even
>> REGISTER with the Asterisk server. - So I can't even try to make a call.
Hi,
I have a Xorcom Astribank connected to my Asterisk server. In one of the
Astribanks FXO port I have a Celular Interface Module. My problem is the
Astribank is receiving a early answer from the module, which doesn't happen
with a ATA connected to the same module. This is causing some trouble wit
On Mon, Jun 16, 2008 at 6:39 AM, voip crazy <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> I have an asterisk PBX working perfectly, and the transfers between
> extensions, works ok. The problem, when I receive a call from the line
> connected to the TE12Xp, and I try to transfer it, the calls hangs u
On Mon, 16 Jun 2008, Gary Guthary wrote:
> If I take a device to my office (i.e. the Sipura) and connect it. - It is
> configured to 'talk' to my home's 'public IP'. - This thing doesn't even
> REGISTER with the Asterisk server. - So I can't even try to make a call.
>
> This is verified (from the
On Mon, Jun 16, 2008 at 7:33 AM, Gary Guthary <[EMAIL PROTECTED]> wrote:
> Hi folks.
>
> Please don't flame me but I've been googling around for days, read a
> tremendous amount, tried everything, and still no go.
>
> This is most definitely a typical newbie question. - I sure hope there's
> somebo
Hi folks.
Please don't flame me but I've been googling around for days, read a
tremendous amount, tried everything, and still no go.
This is most definitely a typical newbie question. - I sure hope there's
somebody(s) out there who'll humble themselves to help me out.
I've set up an 'out of the
More info about the problem.
This occurs, when I try to transfer using the *2 funcionality into aterisk
Thanks
2008/6/16 voip crazy <[EMAIL PROTECTED]>:
> Hello all,
>
> I have an asterisk PBX working perfectly, and the transfers between
> extensions, works ok. The problem, when I receive a ca
Thanks for the link. I think I will be using this product.
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: Saturday, June 14, 2008 1:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [aste
Hello all,
I have an asterisk PBX working perfectly, and the transfers between
extensions, works ok. The problem, when I receive a call from the line
connected to the TE12Xp, and I try to transfer it, the calls hangs up.
I have other analog lines and I can tranfer all the without problems.
I've pa
Having a weird issue with some agents getting stuck busy on my system. Call
will come into the queue and the agent will hit DND, or be DND when the call
comes in (DND being the button on eyeBeam softphone, not a star code).
After the agent comes back from DND they will be "stuck" as busy in the
qu
Sema Arca wrote:
> Can you still send the config files? Maybe I can come up with an idea? :(
extensions.conf entry
exten => _1XX,1,Dial(OOH323/[EMAIL PROTECTED])
exten => _1XX,2,Congestion
ooh323.conf
[general]
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
context=default
tos=lowdelay
di
Can you still send the config files? Maybe I can come up with an idea? :(
On Mon, Jun 16, 2008 at 10:32 AM, Richard Scobie <[EMAIL PROTECTED]>
wrote:
>
> Sema Arca wrote:
> > Hi Richard,
> >
> > I could not succeed to make my ooh323 work somehow. I can see the peers
> > and the users but although
Sema Arca wrote:
> Hi Richard,
>
> I could not succeed to make my ooh323 work somehow. I can see the peers
> and the users but although my exten definition states that the call
> should be forwarded to a GK, Asterisk does not send it out. I also have
> the same problem with registration.
>
>
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