On Fri, 8 Apr 2005, Ronald Wiplinger wrote:
Is priority "n" already supported as next one???
I'm running CVS-HEAD. I don't know when it ('n') appeared.
Steve Edwards wrote:
On Thu, 7 Apr 2005, Jason Brown wrote:
Does anyone have a working failover outbound calls that I could sponge a
hint from? i.e
Hello,
I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have
configured a test account on iax.conf:
[test]
type=friend
context=test
username=test
auth=md5
secret=testing
host=dynamic
disallow=all
allow=ilbc
allow=gsm
callerid=1010
trunk=no
qualify=no
Then I insert an entry on m
On Apr 7, 2005 8:36 PM, kaiser <[EMAIL PROTECTED]> wrote:
> Hi , all:
> Anyone try sip channel with canreinvite=yes?
>
> sometimes we see a new INVITE will be send to UA immediately after user
> hangup the call.
> It makes the phone ring again after hangup.
> Anyone know what happen?
> It not alwa
The configuration for X-Lite in sip.conf:
[177209]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
;regexten=1234 ; When they register, create extension 1234
;username=xlite1
;c
Outstanding!
This completes the usability features of the scheduler. I have
a couple enhancements to make, such a CDR like facility to allow
examining past conferences to see who participated.
For the list members that have been following my app_cbmysql and
Web-MeetMe progress, look for an upda
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%20CallManager%20Express%20Integration
dakemp wrote:
Hi,
We have Asterisk (CVS-HEAD-11/23/04-10:52:47) and Callmanager 4.1(2)
connected together using a SIP trunk, have both way dialing and are using
the Asterisk Box as a voicemai
Hi,
We have Asterisk (CVS-HEAD-11/23/04-10:52:47) and Callmanager 4.1(2)
connected together using a SIP trunk, have both way dialing and are using
the Asterisk Box as a voicemail server for the CCM. Everything is working
great except for the MWI. The CCM has 2 numbers in the MWI config fo
Hi
Il giorno ven, 08-04-2005 alle 10:24 +1200, Matt Riddell ha scritto:
> Matteo Brancaleoni wrote:
> > I hate to say that, but the problem is that Digium doesn't do this.
>
> Ahh I beg to differ.
>
> I resell both Digium and Sangoma gear and provide full installation
> support for both.
after
Correct -- I experience the same with the Sipura ATA's. Certain people
more than others seem to trigger the DTMF detection (especially on the
hs, and hs).
> Switching to in-band fixes it. Well, works around it. :)
That's only an option on G711, I think. Other codecs (G729, G726)
can't tra
> Well I've boiled this problem down to the basics and it is utterly
> repeatable and looks to be an asterisk problem from my ignorant
> point of view.
>
> Basically my asterisk box seems oblivious to incoming DTMF tones
> coming via calls from livevoip via IAX, though it works for
> other SIP pro
Sorry, I should have stated that the position is a FULL-TIME position, based
in our Melbourne office.
- Original Message -
From: "Jean-Michel Hiver" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, April 08, 2005 2:55 PM
Subject: Re: [Aste
Jean-Michel Hiver wrote:
Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users mailing
list? This breaks the reply to sender only / reply to all / list reply
functionality of my mailer. It's really broken :(
Incredibly, some on the
Ronald Wiplinger wrote:
Matt Riddell wrote:
Matt Schulte wrote:
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Once somebody told me, if you do not know what it is, you most likely do
not need it.
However, I can hardly follow that advice. What i
Oops, sorry for the list reply :/
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
---> http://ykoz.net/voip/max <---
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Jean-Michel Hiver wrote:
Oops, sorry for the list reply :/
Actually, why does the Reply-To point to the Asterisk Users mailing
list? This breaks the reply to sender only / reply to all / list reply
functionality of my mailer. It's really broken :(
Cheers,
--
Ykoz Un Max - La VoIP en pré-payé!
Es
Matt Riddell wrote:
Matt Schulte wrote:
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Once somebody told me, if you do not know what it is, you most likely do
not need it.
However, I can hardly follow that advice. What is SRV?
bye
Ronald
_
Rod Bacon wrote:
Sorry if this is off-topic, but I know there's a quite a few smart
people who frequent these groups, and I was thinking that it'd be a
good place to ask.
We have an opening for an experienced PERL programmer. If you (or
anyone you know) is interested, please feel free to email
Sorry if this is off-topic, but I know there's a quite a few smart
people who frequent these groups, and I was thinking that it'd be a good
place to ask.
We have an opening for an experienced PERL programmer. If you (or anyone
you know) is interested, please feel free to email me for more detai
Too bad posts made to the GG do not get mirrored here...
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Calvis
> Sent: Thursday, April 07, 2005 10:11 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [
Has anyone had any experience with the IAX2 phones being marketed by
Netweb?
--
Cory Andrews
Senior Partner
+++
VOIPSupply.com
A Subsidiary of b2 Technologies
+++
voice - 800.398.VOIP X22
fax - 716.630.1548
email - [EMAIL PROTECTED]
___
Matt Loretitsch wrote:
Looking for some help any way I can. I've been closely following
digium's troubleshooting steps and seem to be okay there. I am
connecting, via PRI, to a Definity system. When I release the board on
the Definity side I get this in Asterisk:
*CLI> Apr 7 10:17:23 NOTICE[130
Hello, The Multitech VOIP line supports T38 and I have tested it. It
works great. You will need a public IP to make it work. Very expensive
though. T38 Is not compatible with Asterisk.
Scott Wolfe wrote:
I have been on the same path although I am using a TDM400. No matter what I
did I cou
This has some potential especially for searching.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Thursday, April 07, 2005 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Google
I made that but still same no ringing for pri coming calls
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Mathew McKernan
> Sent: Friday, April 08, 2005 5:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [As
Damon Estep wrote:
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss something?
Looks like a mirror of the mailing list...
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Ast
They are still telling not ringing
Here is the carrier log part
08.04.2005 06:35:10 Connected without receiving ringing, getting call
details on channel 239.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of LJ
> Sent: Friday, April 08, 2005 5:48
Matt Schulte wrote:
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Email for details.
Add this to the bounty page on the wiki (www.voip-info.org)
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news
I want to use the appradius module (specifically the cdr_radius.so)
module to dump asterisk CDR to a RADIUS server (in addition to the local
SQL database). I don't want the authorisation component, only the
CDR->RADIUS function.
I have downloaded, compiled and installed the software without err
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss something?
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRI
If you feel your pots lines are critical to your business then you
should keep them.
What I want is the flexibility and features * gives me, with the
reliability of land lines
Then why do you need a PRI? You can use a channel bank to convert to
PRI if you have DID's. There is a lot you can d
Hi , all:
Anyone try sip channel with canreinvite=yes?
sometimes we see a new INVITE will be send to UA immediately after user
hangup the call.
It makes the phone ring again after hangup.
Anyone know what happen?
It not always, maybe 2-5% only.
But it make user crazy.
Thanks...
_
Hi all,
I have a VoIP PBX box with asterisk and one x100p card. I setup some sip users in sip.conf.
The asterisk will quit aperiodically, sometime it will work for several days before quit, but I find its quit time is almost in 18:00 to 19:00.
I can not find any clue from log file. The
Under what type of relationship? Are you a CLEC, ITSP, or Retail
Customer? Is the local carrier you mention the incumbent or a CLEC?
Thanks
>
>
> One of our local carriers charges 17 cents per ported DID MRC, no
port/non
> recurring charges.
>
> I've seen in the neighborhood of $15 per 10 porte
Are you asking or saying it can be done that way?
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Thursday, April 07, 2005 11:56 AM
To: Asterisk Users Ma
hello
how to pass G723.1 to other side is there any
softphone using g723.1. i want to use G723.1 in my
voice communication.
regrads
Kamran
__
Do you Yahoo!?
Yahoo! Personals - Better first dates. More second dates.
http://personals.yahoo.com
On Apr 7, 2005 9:30 AM, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Matteo Brancaleoni wrote:
>
> > ARA is something different from ast_data:
> > * ast_data is real realtime: for example extensions.conf
> > is looked up from DB in realtime without reload.
>
> Hmm. RealTime extensions are exa
I had a similar problem where the PBX I was connecting to would not
recognize the answer until I set Ringing() before the answer. I do not
recall if I used a wait in between. It was something like:
Exten => 2688,1,Ringing()
Exten => 2688,2,Wait,1
Exten => 2688,3,Answer
Hope that helps.
--LJ
--
Steve Edwards wrote:
On Thu, 7 Apr 2005, Jason Brown wrote:
Does anyone have a working failover outbound calls that I could sponge a
hint from? i.e.
Exten => _1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten =>
_1NXXNXX,2,Dial(IAX/IfMyFirstCrappyIAXPeerDontAnswerIn5SecDialMySecond
I'd try turning the RXGAIN and TXGAIN down some... they're probably what's
giving you the excessive side-band audio.
Ian
>>> [EMAIL PROTECTED] 07/04/2005 16:38 >>>
Hello everybody,
my setup consists of an asterisk server with a TDM400P and a couple of
softphones (SJphones) ... everything wor
On Apr 7, 2005 6:32 PM, Michael D Schelin <[EMAIL PROTECTED]> wrote:
> I may be able to help. I'm a provider in Southern CA. What you need to
> do is eliminate all pots lines by moving them over to VOIP completely.
> This will take some time but will save you a lot of money. Please call
> me for m
Message: 6
Date: Thu, 7 Apr 2005 16:24:18 -0700
From: snacktime <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Getting a good deal on a PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
We have 10 i
On Thu, 7 Apr 2005, Jason Brown wrote:
Does anyone have a working failover outbound calls that I could sponge a
hint from? i.e.
Exten => _1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten =>
_1NXXNXX,2,Dial(IAX/IfMyFirstCrappyIAXPeerDontAnswerIn5SecDialMySecondCr
appyIAXPeer
Exten
Message: 6
Date: Thu, 7 Apr 2005 16:24:18 -0700
From: snacktime <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Getting a good deal on a PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
We have 10 i
Hi all,
I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok except there is a short audio latency from PSTN to SIP and no delay in the reverse direction.
At the beginning of a call, the latency is not very long, but it becomes more and more obvious along with time.
Does anyone have a working failover outbound calls that I
could sponge a hint from? i.e.
Exten =>
_1NXXNXX,1,Dial(IAX/MyFirstCrappyIAXPeer/${EXTEN},60)
Exten => _1NXXNXX,2,Dial(IAX/IfMyFirstCrappyIAXPeerDontAnswerIn5SecDialMySecondCrappyIAXPeer
Exten =>
_1NXXNXX,3,Dial(IA
Hi,
Where you have your 1st priority, I suspect you have it set to "Answer".
Try changing this to Wait(1). Then on priority 2 put answer. i.e.
Exten => s,1,Wait(1)
Exten => s,2,Answer
Exten => blah blah
Hope that covers it,
Thanks
Mathew
-Original Message-
From: [EMAIL PROTECTED]
[ma
Hello, I've recently been testing egressing calls from asterisk via SIP
to a Max TNT (TAOS 10.1.1). I'm able to route the calls to it and
complete the calls successfully, however the called-party id gets lost
somewhere. In my log output on the TNT I see.
When the call is placed I see, the called-f
On April 7, 2005 09:38 pm, Ugur GUNCER wrote:
> How can i set asterisk for when call came from pri ring once then answer
> pri call.
>
> In now call cames from pri then asterisk directly answering pri call
> without ringing. Then my carries hangup call because they said your box is
> answer without
Hello everyone, I need to configure a stand alone Voice mail box. Calls
will come in via sip. I have read and read until my eyes hurt for 2
weeks now. Can someone email me the basic config files needed to do
this. The examples are overly complicated. I just need a simple basic
configurations wi
Brandon Patterson,
Okay at this point it should be know that Livevoip. Does not support
DTMF over IAX. Why not save the time and trouble and stop selling Level
3 DID's?
After all the trouble that you have had with Level 3 DID's why would you
even sell an unstable product? Like this.
On Thu, 20
Hi all,
Due to the advancement of a particular project & crushing time frames
associated with it, we need to look at hiring a consultant to help
accelerate a particular implementation.
Does anyone know of someone for hire who would be interested in such a
project ?
If so, please feel free to c
How can i set asterisk for when call came from pri ring once then answer pri
call.
In now call cames from pri then asterisk directly answering pri call without
ringing. Then my carries hangup call because they said your box is answer
without ringing
Iyi Calismalar
Saygilarimla
Ugur GUNCER
Hello,
Here's jsut a simple manager Action to send, make sure that you have an
extension set up to play the message(exten => 1234,1,Playback(file)) and
that's the extension that will be called from the meetme room. Also, make
sure that that extension calls in to the meetme room extension with the
I may be able to help. I'm a provider in Southern CA. What you need to
do is eliminate all pots lines by moving them over to VOIP completely.
This will take some time but will save you a lot of money. Please call
me for more info. I can provide you service and if your interested
LNP your nu
Grandstream has the same problem. Very common. A simple DTMF debouncer
curcuit will fix it.
Doug Meredith wrote:
Eric Wieling <[EMAIL PROTECTED]> wrote:
Daryll Strauss wrote:
Yep, I've seen it and from reading http://www.voxilla.com it's a
pretty common problem.
I use the attached AGI script (place it in your agi-bin directory). It
places a call file in asterisk's outgoing spool directory
(/var/spool/asterisk/outgoing) in my case.
The script is called with as argument the number to call back. Note that
if the call back number is 5 digits or less, it actu
LiveVoip Supports the every changing and improving Asterisk Code
for many many customers on a daily basis. In the case of the DTMF
issue we have people working on it. No estimated time to a solution. The
work continues. This looks like you are one of the 5% that we may not be
able to support. So we
Hello all,
I am working with an AAH installation and Polycom IP 500 phones. Phones
are now working and I'm just trying to fine tune what settings I need in
my extension and sip .conf files. I have AMP installed obviously (its
AAH) but I am finding that I will probably not use the AAH extension
On Thu, 2005-04-07 at 15:17 -0700, Dan Austin wrote:
> I am wrapping up a PHP addon script to my scheduling
> framework and have it properly tracking and closing
> conferences.
>
> I need to play an announcement into the room that the
> conference will end soon. I haven't found a great way
> to d
Eric Wieling <[EMAIL PROTECTED]> wrote:
>Daryll Strauss wrote:
>
>> Yep, I've seen it and from reading http://www.voxilla.com it's a
>> pretty common problem.
>>
>> If you turn on debugging what you'll see is that the Sipura has
>> mistakenly detected a DTMF code in the audio stream and is relayi
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003976.
Essentially, the bug is that if a callback agent puts a caller on hold,
that caller does not hear MOH. This bug has been around for a while,
but nobody has been able to follow through on testing to the point where
we could na
snacktime wrote:
We have 10 incoming POTS lines to our offices, and a nortel norstar
pbx. I've been looking at replacing it with * at some point in the
future, and the point that looks most cost effective is when we move
to PRI.
Problem is, I'm not really sure how to go about getting a good deal,
PRI pricing really depends on where you are at.
It's best to talk with the CLECs in your area, I have done quite well
with TW Telecom in Tucson AZ.
I have used TW Versipak, which they do as a PRI T1 (fractional, of
course) voice hand over, and Ethernet Data handover. Normally becomes
competative
Oh.. the voltage values are dumped in syslog!
Leandro Morgado wrote:
Damian Minkov wrote:
Is there a way to measure the signal of the connected line on the FXO
port ( without the help of digital oscilloscope )
Yes there is. But you need to edit the source code of wcfxs (for the
TDM400 card). The
We have 10 incoming POTS lines to our offices, and a nortel norstar
pbx. I've been looking at replacing it with * at some point in the
future, and the point that looks most cost effective is when we move
to PRI.
Problem is, I'm not really sure how to go about getting a good deal,
or what question
A sample would be great. I'm hoping that the Official MeetMe2
will have provisions for this, but until then I'll have a
fully functional scheduler.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: Thursday, April 07, 2005 3:31 PM
To: 'Ast
One of our local carriers charges 17 cents per ported DID MRC, no port/non
recurring charges.
I've seen in the neighborhood of $15 per 10 ported numbers as an "LSR"
charge from other carriers NRC.. and as low as 5 cents MRC per Month.
I've also seen cases with no MRC per DID per month, but an N
Damian Minkov wrote:
Is there a way to measure the signal of the connected line on the FXO
port ( without the help of digital oscilloscope )
Yes there is. But you need to edit the source code of wcfxs (for the
TDM400 card). There is a bit of code similar to:
#if 0
-->some debug messages here re
On Thu, Apr 07, 2005 at 10:23:50PM +0100, Gavin Hamill wrote:
> On Thursday 07 April 2005 22:17, Alex Vishnev wrote:
> > Magnus,
> > Also, compression gives voice recognition quite a challenge,
> > as the speech samples arriving at the voip voice recognition engine is not
> > the same as it was sp
Anyone out there (in the US) using a CLEC to do third party local number
ports? Let me be more specific;
Our inbound calls come in via inbound only PRIs from a local CLEC, our
outbound calls go via SIP termination to a wholesale VoIP carriers
softswitch.
On the inbound numbers we use the carrier
My voice messages emailed to me have poor volume, to the
point where I can barely hear them. I have confirmed by loading them in
Audition that the level is poor. This happens on PSTN and VOIP calls. Recording
my own voice messages using the record application results in very low levels
also
As long as this competition is setup to be delivered after July I should
be able to donate a www.akimbo.com box to the winner (I'm going to be
distributing them in Australia).
You will still need to pay for the content and the monthly charges but
it's still a kick arse prize.
Cheers,
Dean
---
Hello Victor:
Are you using the QuadBRI from Junghanns?
If so, this is a configuration that works for me in the UK, one single
E1 board and the QuadBRI. As I recall, the order of loading is
important (see way below):
ZAPTEL.CONF
loadzone=nl
defaultzone=nl
# qozap span definitions
# most of the
Well, then go to a psychologist as well. You should not enable this on a list.
On Apr 7, 2005 6:35 PM, Huddleston, Robert <[EMAIL PROTECTED]> wrote:
> Out on medical leave - I will return Monday 4/11/05
>
___
Asterisk-Users mailing list
Asterisk-Users@li
I have 3 FC3 systems in production, of those 2 are giving me some
trouble. I have since decided to stay away from FC3.
However, most ppl don't have any trouble. Some tech support ppl at
Digium use Fedora Core (I think 2).
On Apr 6, 2005 9:00 AM, Altus Snyman <[EMAIL PROTECTED]> wrote:
> Thanks for
just create an extension that plays the message and hangs up and use the
manager interface to drop it into the meetme room.
Let me know if you would like an example and I'll whip one up.
We do this kind of thing in astGUIclient to play DTMF tones automatically in
meetme rooms.
MATT---
-Ori
Hello,
I have a machine with two cards
installed, one digium that gives e1 connectivity and one quadBri for the ISDN
line.
I can use them
independently. I have one zaptel.conf and one zapata.conf for each card. I
would like to work with them at the same time and I am not sure about how
I currently use another PBX system which takes care of VM. Is there a
way to prevent [EMAIL PROTECTED] v0.6 from picking up Incoming calls?
I'd still like to dial out from Asterisk (I have IAX trunking on). Is
there a way to do this? My knowledge of the Extensions.conf is
limited.
I'm using [EM
Matteo Brancaleoni wrote:
I hate to say that, but the problem is that Digium doesn't do this.
Ahh I beg to differ.
I resell both Digium and Sangoma gear and provide full installation
support for both.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/
I am wrapping up a PHP addon script to my scheduling
framework and have it properly tracking and closing
conferences.
I need to play an announcement into the room that the
conference will end soon. I haven't found a great way
to do that. One way that I have thought of, but would
like to avoid is
As requested, here is my database setup
++--+--+-+-++
| Field | Type | Null | Key | Default | Extra
|
++--+--+-+-++
| id
I've seen this or
similar questions in the archives, but haven't seen a solution
yet.
I'm trying to use an
FXO card to connect to a PBX (Toshiba DK-424). Currently I'm testing with
an x100p, but eventually I'll be using TDM400s with FXO & FXS modules.
The Asterisk box is not detecting d
Hello again, many thanks for the feedback, very interested in Dean's
comments that there could be work in progress else where in the world. Are
the other features related to how the personnel directory could be
viewable/recordable via web interface? This would seem worthwhile, also it
would be good
>Cisco unfortunately doesn't care too much about interoperability
>as they prefer to sell their own products..
which, wether you like it or not, are super (PIX not
included).
I've been playing around with a lot of hardware from
different vendors, but Cisco is still on my 'expensive but
works grea
Can this also be in categories.
I have a CRM application that uses asterisk in several ways.
But we are by no means a configuring program. Yet I still
think we are one of a kind.
Since [EMAIL PROTECTED] did a marvelous job in making an
all-in-one ready to go project for a dedicated pbx, my
project
Jan Johansson wrote:
Is it possible (How complicated is it?) to do this;
IVR plays the usual “please type your order number, finish with pound”
Then I would like to query a MSSQL database server, looking up the
“Status” column from a row where ordernr = the entered order number.
Depending on the
On Thu, Apr 07, 2005 at 12:54:21PM -0500, Gabriel Millerd wrote:
> I have been struggling with oh323 compilation for some time now. I am
> trying to use the voip-info suggested walk through that points to here
> ...
>
> http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artl
I have added him, and Joshua Chessman to my filter-to-trash rules. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of dean
collins
Sent: Thursday, April 07, 2005 4:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: FW: Out of Office Au
[EMAIL PROTECTED] wrote:
Hello,
I was just wondering if there were a prize like the open source application of
the year relative to Asterisk?
All these developer doing good job and all free need some present sometime that
we can all donate.
Anything like that exists?
Not yet. But, it wouldn't be i
Haha, this is trueit reminds me of my loud obnoxious car audio days,
competing in the dB drag events
I remember people telling me how they had to spend thousands of dollars to
double their amplifier wattage just to get 3 more dB on the competition.
It's funny how now when some guy pull
Hi All
How can i made conference first person coming from PRI and second person
dialed from asterisk with SIP.
How will be my extension conf
I wrote extension for first person
exten => _5463XX,1,Answer
exten => _5463XX,2,MeetMe(1234|a);
exten => _5463XX,3,Hangup
But i dont know ho
On Thursday 07 April 2005 22:17, Alex Vishnev wrote:
> Magnus,
>
> Also, compression gives voice recognition quite a challenge,
> as the speech samples arriving at the voip voice recognition engine is not
> the same as it was spoken using regular 64kbits pstn connection (as an
> example).
>
If htt
--- "trixter http://www.0xdecafbad.com";
<[EMAIL PROTECTED]> wrote:
> This script does mp3-> wav -> gsm, if you omit the
> first part it should
> work for you. requires mpg123 and sox for wav->gsm
> conversion and
> mp3-decoder to start from mp3.
>
>
> will process all .mp3 files in the current
Magnus,
As far as I remember, Festival is only Text-to-speech, not voice
recognition. In order to do what you want you need a voice recognition
application. Also, compression gives voice recognition quite a challenge, as
the speech samples arriving at the voip voice recognition engine is not the
s
That's not what he is looking for though, this will only dial 1 number
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josiah
Bryan
Sent: Thursday, April 07, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-U
Thanks Robert, that information is really handy to know - now turn the
freaking out of office message off!
-Original Message-
From: Huddleston, Robert [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 4:57 PM
To: dean collins
Subject: Out of Office AutoReply: [Asterisk-Users] V
On April 7, 2005 04:42 pm, Tony Mountifield wrote:
> OK, I'd been told this chip could support both 3.3V and 5V, but from what
> you're saying, it sounds like it can be set up to support 3.3V OR 5V, but
> not both at once officially. Of course, when selling product it is
> prudent only to work
That effort is underway here at LiveVoip as well. *22 drops you into the
Voice Section. Its going to be a paid for offering in Mid May.
Brandon Patterson
Hello all, rumours reach me of a way that the UK incumbent operator is
planning to compete with VOIP by offering voice activated dialling, e.g.
Mike Dewey wrote:
>I am very interested in what you came up with for a 2.5mm
> to RJ-10 adapter.
Yes, I built a 2.5mm cellphone headset to Cisco 7960 RJ10 headset jack
adapter. If I follow the numbering system used on
http://www.mml.uni-hannover.de/einhorn/headset/index_e.html, this is how
it
On Thu, Apr 07, 2005 at 09:08:22AM -0500, Parker, Blake (MIS) wrote:
> I cannot get spandsp to compile. Can someone email me the compiled app
> files for linux.
http://packages.debian.org/unstable/libs/libspandsp0
http://packages.debian.org/unstable/libs/libspandsp-dev
;-)
--
Tzafrir Cohen
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