On 7/17/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Ein Bielaczyc wrote:
>
> > I have a small customer looking to update their aged telemarketing
> > system. I ran across astGUIclient and Vicidial
> > (http://astguiclient.sourceforge.net/) during a Google search and was
> > wondering if anyo
Matt Florell wrote:
> No apologies necessary, I think a lot of what you said is mostly true.
Well, thank you. I really appreciate that you're willing to entertain
what I am saying without construing it as some sort of attack; it is
not in the least bit intended that way.
> The PHP and Perl
Once again it's almost Friday and at 12 Noon EDT on Friday July 18th, 2008
THIS WEEK: Are you in any user groups? If so, what benefits are there?
Digium's John Todd has a tough job, deciding how to help users groups
worldwide to higher visibility while these groups can help raise
asterisk awarene
There is an interesting click to call badge in development which I've
temporarily placed here:
http://bit.ly/click2call
just below the Google calendar.
During tomorrow's conference at Noon EDT on Asterisk users' groups,
you can try it to connect. It would be of interest to several of us
how thi
2008/7/16 Tilghman Lesher <[EMAIL PROTECTED]>:
> > If they TRULY see themselves as a TELCO replacements for large shop
> > they REALLY need to step up to
> > proving INFO, WARN, ERROR messaging in a unified reliable manner. Such
> > as a SNMP messaging ability for all
> > INFO, ERROR, and WARN le
I am forwarding this search for two asterisk systems people to fill a
position in NYC in case someone here is interested:
===
Looking for a potential match for my Asterisk System Architect requirement.
If you are not i
On Wed, Jul 16, 2008 at 3:07 AM, Gordon Henderson
<[EMAIL PROTECTED]> wrote:
> On Tue, 15 Jul 2008, Matt Darnell wrote:
>
>> Does anyone know of a bandwidth test that tests the upload with the download?
>>
>> All of the ones I can find will test the upload then the download.
>>
>> I from experience
Hi,
I'm testing using the free g723 codecs and i have successfully installed
them.
g723
g723 -
gsm 9
ulaw 9
alaw 9
g726 9
adpcm 9
slin 8
lpc1010
g72910
speex -
ilbc10
i also set my pap2's to use G723
Hi All,
Does anyone know, if there is a tool, which is doing the follwing:
- Testprogram on host A establishes a sip connection to testprogramm on
host B
- Testprogram on host A plays a tone and Testprogram B verifies, if tone
is playing correctly (without any interruptions)
Thank You.
__
17 jul 2008 kl. 12.08 skrev Dennis Brandenburg:
> Hi All,
>
> Does anyone know, if there is a tool, which is doing the follwing:
>
> - Testprogram on host A establishes a sip connection to testprogramm
> on
> host B
> - Testprogram on host A plays a tone and Testprogram B verifies, if
> tone
> 17 jul 2008 kl. 12.08 skrev Dennis Brandenburg:
>
>
>> Hi All,
>>
>> Does anyone know, if there is a tool, which is doing the follwing:
>>
>> - Testprogram on host A establishes a sip connection to testprogramm
>> on
>> host B
>> - Testprogram on host A plays a tone and Testprogram B verif
Artie Gold wrote:
> Is there some way to specify the use of a different codec (in this case,
> g.711 vs. g.729) for use with meetme?
> We have found that in the meetme case, with the necessity of
> decompression and recompression, it's worth it to us to use g.711 (use
> the bandwidth and save the p
Steve Underwood wrote:
> You might think a standard phone plugged into an adaptor, like a
> Magic-jack, would be limited to narrow band voice, as that is all the
> phone was designed for. It turns out most phones only aggressively
> filter at the low end of the band. They let a lot of energy ab
On July 16, 2008 08:01:38 am Loic Didelot wrote:
> Hello,
> I would like to double check what Echo Cancellation my Digium Card uses.
>
> I thought I bought the little more expensive card that integrates
> EchoCancellation. How can I check?
If you load the modules with debug=1 (maybe this appears
HPEC i see also someone who correct the C code to cancel echo you can
find that on google
2008/7/16 Loic Didelot <[EMAIL PROTECTED]>:
> Hello,
> I would like to double check what Echo Cancellation my Digium Card uses.
>
> I thought I bought the little more expensive card that integrates
> E
17 jul 2008 kl. 12.51 skrev Dennis Brandenburg:
>
>
>> 17 jul 2008 kl. 12.08 skrev Dennis Brandenburg:
>>
>>
>>> Hi All,
>>>
>>> Does anyone know, if there is a tool, which is doing the follwing:
>>>
>>> - Testprogram on host A establishes a sip connection to testprogramm
>>> on
>>> host B
>>> -
Yes, it's an _X. match for local/ld
It actually ended up being oddity with Centos 5.2, I had to upgrade Zaptel to
the newest version and it resolved it, apparently it wasn't passing all the
digits to the line.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behal
On Thu, Jul 17, 2008 at 02:31:03PM +0200, Johansson Olle E wrote:
> I would love for someone to develop that application for Asterisk. It
> should be doable,
> but is way out of my knowledge.
>
> The Spirent people said they where sending a "common" three tone
> sequence and
> tested what cam
I have an opportunity to interface asterisk with a security system to
open their magnetic door locks. The security system needs a dry contact
close upon activation to signal the door. Has anyone done this before?
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Quoting c james <[EMAIL PROTECTED]>:
yep, and now there are even channel banks that have contacts built right in.
just google it.
> I have an opportunity to interface asterisk with a security system to
> open their magnetic door locks. The security system needs a dry contact
> close upon activa
Hi David,
It may be IAX2 bug, do you use IAX? In my case downgrading back to 1.4.19
did the job.
Michael
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://w
I think you can buy some kind of ATA's to do such a job.
I do not however remember any brand names
Google returned these links:
http://www.voip-info.org/wiki/view/Asterisk+phone+door&view_comment_id=15775
http://www.abptech.com/products/its.html
Mobotix
c james wrote:
> I have an opportunity to
Right. I fully understand that. The preference for g.711 in this case is
that the transcoding is significantly cheaper (as reflected in the
information obtained from "core show translation").
Thanks,
--ag
On Thu, Jul 17, 2008 at 6:45 AM, Kevin P. Fleming <[EMAIL PROTECTED]>
wrote:
> Artie Gold
Hi James
I did it some years ago to open an electric door entry. I used an
home-made board with embedded tcp-ip stack and a perl AGI scripts that
sends some UDP packets containing a "secret" passphrase.
The AGI it's still triggered by an internal extension call.
I think that today you can find so
17 jul 2008 kl. 15.08 skrev zoa:
> I think you can buy some kind of ATA's to do such a job.
> I do not however remember any brand names
>
> Google returned these links:
> http://www.voip-info.org/wiki/view/Asterisk+phone+door&view_comment_id=15775
> http://www.abptech.com/products/its.html
> Mobo
Hi Loic -
> According to that its using MG2.
I think it will say MG2 regardless of whether or not there is a
hardware module present.
> Shouldnt it be using something like
> HPEC?
I don't think the hardware echo cancellers use the HPEC algorithm. As
Eric and Matt have mentioned, dmseg will te
Artie Gold wrote:
> Right. I fully understand that. The preference for g.711 in this case is
> that the transcoding is significantly cheaper (as reflected in the
> information obtained from "core show translation").
So then I don't understand your question... are you asking for some way
to tell t
Kevin P. Fleming wrote:
> Steve Underwood wrote:
>
>
>> You might think a standard phone plugged into an adaptor, like a
>> Magic-jack, would be limited to narrow band voice, as that is all the
>> phone was designed for. It turns out most phones only aggressively
>> filter at the low end of t
On Wed, Jul 16, 2008 at 08:33:44PM -0400, Ein Bielaczyc wrote:
> I have a small customer looking to update their aged telemarketing
> system. I ran across astGUIclient and Vicidial
> (http://astguiclient.sourceforge.net/) during a Google search and was
> wondering if anyone had any experiences to s
Yes. Exactly.
Thanks!
--ag
On Thu, Jul 17, 2008 at 9:01 AM, Kevin P. Fleming <[EMAIL PROTECTED]>
wrote:
> Artie Gold wrote:
> > Right. I fully understand that. The preference for g.711 in this case is
> > that the transcoding is significantly cheaper (as reflected in the
> > information obtaine
On 7/17/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Matt Florell wrote:
>
> > No apologies necessary, I think a lot of what you said is mostly true.
>
>
> Well, thank you. I really appreciate that you're willing to entertain
> what I am saying without construing it as some sort of attack; i
Artie Gold wrote:
> Yes. Exactly.
>
> Thanks!
> --ag
>
If you're using SIP phones, you can set the SIP_CODEC channel variable in the
dialplan prior to the call to MeetMe. If it's G.711 you want, set SIP_CODEC to
either "ulaw" or "alaw."
As for other technologies, I don't know of a way to over
Perfect! (Yes, we're using sip phones).
Many, many thanks!
--ag
On Thu, Jul 17, 2008 at 9:38 AM, Mark Michelson <[EMAIL PROTECTED]>
wrote:
> Artie Gold wrote:
> > Yes. Exactly.
> >
> > Thanks!
> > --ag
> >
>
> If you're using SIP phones, you can set the SIP_CODEC channel variable in
> the
> dial
Artie Gold wrote:
> Perfect! (Yes, we're using sip phones).
That won't change the codec in use for an existing channel. Once the
channel has been answered Asterisk has no method (currently) to change
the codec itself, although it will respond to a codec change initiated
by the remote endpoint.
--
On 7/17/08, Matt Florell <[EMAIL PROTECTED]> wrote:
> On 7/17/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
> > Matt Florell wrote:
> >
> > > No apologies necessary, I think a lot of what you said is mostly true.
> >
> >
> > Well, thank you. I really appreciate that you're willing to enter
I believe HPEC actually is the same algorithm (G.168) that the HW echo
cancel modules use.. the difference being that HPEC uses up CPU cycles and
its performance will be impacted on a system with higher CPU load, whereas
the HW modules have a dedicated DSP for it.
http://blogs.digium.com/2007/09/0
On 7/17/08, Steve Underwood <[EMAIL PROTECTED]> wrote:
>
> That is certainly true. Many of the comments people make about codecs
> owe more to the phone than the codec. However, there are various types
> of impairment. Even a phone which doesn't sound nearly as good as it
> could in G.711 mode m
1/ R&D costs v's number of units manafactured per annum.
2/ Retail pricing the markets will still purchase at.
And yes you are right the polycom's are too expensivebut budgetone's
are still too expensive even at $20 :)
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mail
On Thu, Jul 17, 2008 at 07:43:44AM -0500, c james wrote:
> I have an opportunity to interface asterisk with a security system to
> open their magnetic door locks. The security system needs a dry contact
> close upon activation to signal the door. Has anyone done this before?
Worst case, Ciarci
On Thu, Jul 17, 2008 at 11:34:30AM -0400, Kristian Kielhofner wrote:
> Could someone please explain to me why business desk phones are so
> expensive? I'm not knocking my friends over at Polycom, Snom, or any
> other manufacturer but in some cases you can buy a cheap but usable
> laptop for less
Matt Watson wrote:
> I believe HPEC actually is the same algorithm (G.168) that the HW echo
> cancel modules use.. the difference being that HPEC uses up CPU cycles
> and its performance will be impacted on a system with higher CPU load,
> whereas the HW modules have a dedicated DSP for it.
G.16
Dean Collins wrote:
> 1/ R&D costs v's number of units manafactured per annum.
>
The only business phone I ever contributed to had a run rate of about
500K/annum and a production life of multiple years (not sure how many it
lasted for). I put some (at the time) exotic DSP into a high end
prod
I'm about to go off and try to write a script that parses the output of
show channels concise so that I can get something readable (since show
channels doesn't help me much either)...
unless someone else has already done somethign similar and wants to
share?
Cheers,
-- jra
--
Jay R. Ashworth
On Tue, Jul 15, 2008 at 04:43:44PM -0400, John covici wrote:
> > http://en.wikipedia.org/wiki/?-law_algorithm
>
> OK, thanks -- I was a bit puzzled because if I want to play audio over
> asterisk, it has to be 16-bit 8khz signed, so that is what I thought
> ulaw was -- thanks for the clarificatio
Quick Question...
I'm trying understand, if it's possible to run an agi script to obtain a
user's account balance and from there asterisk would be able communicate
that value back to a sip phone. Is that phone feature, or is that an
asterisk feature already?
Think of this as on a prepaid pla
Matt Florell wrote:
> But seriously, Asterisk is a better example of doing things right more
> recently, a couple of years ago all sorts of stuff went into the
> "stable" releases of Asterisk without enough testing resulting in
> some pretty big bugs (like in asterisk 1.2.10-12) And even further
Gerard A. Matthew wrote:
> I'm trying understand, if it's possible to run an agi script to obtain a
> user's account balance and from there asterisk would be able communicate
> that value back to a sip phone. Is that phone feature, or is that an
> asterisk feature already?
Phone feature. You
On Thu, 17 Jul 2008, c james wrote:
> I have an opportunity to interface asterisk with a security system to
> open their magnetic door locks. The security system needs a dry contact
> close upon activation to signal the door. Has anyone done this before?
Xorcom have some nice kit... Shame it co
Hi what version of openh323 and pwlib are suggested for asterisk
1.4.21.1.? Thanks to all
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
___
On Thu, 2008-07-17 at 19:34 +0200, nik600 wrote:
> Hi what version of openh323 and pwlib are suggested for asterisk
> 1.4.21.1.? Thanks to all
Iirc it is openh323 1.18.0 and pwlib 1.10.1.
Regards,
Patrick
___
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Hi,
I have a client that wants the following scenario using asterisk. I have
done a similar and smaller scenario with Cisco Call Manager, but being an
asterisk beginner, I don't think I can do it. So I needed some basic hints
about each part of the project. Maybe a line or a word saves me a lot of
Mark Mickan wrote:
> I've tried configuring the Asterisk end with em_w, featb, featd, featdmf
> but with each of these, it either doesn't work at all, or I see calls
> coming in to Asterisk that shouldn't be, and unexpected robbed bit
> patterns at the simulator end.
Can you post what combinat
I can't transfer calls with my polycom 501's. Do I need to set up
something in particular in the asterisk dialplan to make the feature work?
___
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AstriCon 2008 - September 22 - 25
How about something like this?
http://www.pencomdesign.com/1ch_relay.htm
No personal experience with that board, but it looks sensibly designed and
should be very easy to code for.
-Keith
- Original Message -
From: "c james" <[EMAIL PROTECTED]>
To:
Sent: Thursday, July 17, 2008 5:43
Was it like watching a 106' plasma at 1080p for the first time?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, July 17, 2008 7:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
> I know you cannot describe the whole scenario in an email, what I need
> is a line or some words for each step :) Or if anyone can do the
> whole scenario, please send me an email for further discussions.
>
Would a combination of call files and conference rooms achieve this?
Quick thoughts
I'm trying to write an application for using after an agent has decided the
person on the other end is an answering machine and would like to drop in a
message automaticly. When I'm testing this using my own voice as an
aswering machine, WaitForSilence works correctly and returns only after a
dece
This is what we use, with (seemingly) good success:
exten => answermachine,1,Answer
exten => answermachine,n,Wait(5)
exten => answermachine,n,WaitForSilence(1000,2)
Julian
Nicholas Blasgen wrote:
> I'm trying to write an application for using after an agent has decided the
> person on the o
My Zoom 5801 ATA hangs up at 30 seconds every call.
I do not think it´s an Asterisk issue, as calls on the SIP trunk goes in and
out normally.
Below is the CLI message.
216 is the extension number assigned to the FXS extension port on the ATA.
Another problem that came up while I was trying to s
I think it should work standard (i.e. no special setup) Do you have a
transfer button on the phone?
Kate
Adam Moffett wrote:
> I can't transfer calls with my polycom 501's. Do I need to set up
> something in particular in the asterisk dialplan to make the feature work?
>
>
>
>
thanks for your reply.
I've installed them but i'm experiencing this problem:
i've configured in h323.conf 2 peers:
one to an 3.3 CCM Cisco
one to an 4.2 CCM Cisco
each CCM has the preferred codec set up as G711 ulaw.
I can forward calls from a SIP account on asterisk (using Xten-xlite
as softp
Thanks for responding Kate.
I do have a transfer button on the phone, and I follow the transfer
process as described in the user's guide. When I press "transfer" the
first caller is placed on hold and then I call the party I want to
transfer to. At this point I'm supposed to press "transfer"
Any thoughts on how to relate the call from Box 1 to the original call ?
On Thu, Jul 17, 2008 at 10:58 PM, Bob Pierce <[EMAIL PROTECTED]> wrote:
>
>
> > I know you cannot describe the whole scenario in an email, what I need
> > is a line or some words for each step :) Or if anyone can do the
>
Hi There,
We are using 2 x AVM Fritz BRI cards with mISDN. The phones are
Linksystem SPA922's and we are getting a little echo on the lines..
from what i unserstand, these are passive cards and do not have any
onboard echo cancellation, but im wondering if there is anything that
can be done softwa
Adam,
We have the exact same issue occurring on one of our networks. I haven't had
time to dive into it much, but here is what I've found out:
Calling from a softphone via IAX2 or SIP, transfers work fine.
Calling from a Polycom 501 outside the network, transfers work fine.
However, the Polycom I
Generally, when you see a call always hang up at 30 seconds it is
because you are not "answering" in your dialplan before doing other
things.
As for the reset, you may want to hold in the reset button for like 30
seconds, pull the power plug and plug it back in after 10 seconds
while holding down
One of the last "secure" facilities I worked in had a motion sensor
that unlocked the door for people leaving from one door. The COO was
pretty shocked when I took off my belt, easily pushed it through the
gap in the glass doors near the top and triggered the motion sensor,
immediately opening the
Not dialing a 1? Make sure it as actually send the call out the zap
interface. Whats the console say? :)
Jeremy Mann wrote:
>
> Can anyone help me start to diagnose why a Sangoma A200 wouldn’t dial
> out LD? Local calls are fine, incoming is fine, just no LD. Bell tech
> has been on site and pl
> One of the last "secure" facilities I worked in had a motion sensor
> that unlocked the door for people leaving from one door. The COO was
> pretty shocked when I took off my belt, easily pushed it through the
> gap in the glass doors near the top and triggered the motion sensor,
> immediately o
Here are other solutions
http://www.abptech.com/blog/open-doors-with-sip/
http://www.netgenium.co.uk/documents/ip_lock_controller.html
http://www.premierelect.com/10.cfm?prodCode=1031&category=88
-E
http://mobiquity.ws
http://gpro.ws
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Yes, I currently work for a Hosted Service provider - we have a cage
in a datacenter with T1s running into it. We find it a nightmare
dealing with hardware issues, power outages, ultimately affecting our
customers. It takes a lot of resource to manage the cluster and keep
it up and running. Its
Hi everyone,
I`ve been having several problems with my current softphone and I`m trying
to develop an IAX Client based one.
Does anyone know how can I get help or useful resources about it? Specially
with Conference function and management of incoming call events to launch an
AGI at that time.
I´
On 7/17/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Matt Florell wrote:
>
> > But seriously, Asterisk is a better example of doing things right more
> > recently, a couple of years ago all sorts of stuff went into the
> > "stable" releases of Asterisk without enough testing resulting in
> >
I remember driving past a building with a magnetic door lock (where a
friend worked) late one night, and he noticed that their door was open
and swinging in the breeze. Turns out that if you lost power, this
particular door would open up as well.
PaulH
Steve Totaro wrote:
> One of the last "
Paul Hales wrote:
> I remember driving past a building with a magnetic door lock (where a
> friend worked) late one night, and he noticed that their door was open
> and swinging in the breeze. Turns out that if you lost power, this
> particular door would open up as well.
>
> PaulH
Heck, that
On July 17, 2008 11:44:07 am Dean Collins wrote:
> 1/ R&D costs v's number of units manafactured per annum.
That's bullshit; There are many more office phones than office desktops out
there, and the research has been paid for many times over. Think of how long
the Meridian 1 has been around.
>
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