I've looked into this. The important reasons as to 'why this shouldn't
happen' are:
Requires a Database - (bad for flash, also very large)
Needs apache + php (+30 odd mb)
A fair whack of perl modules (+10mb)
== Too large, too cumbersome.
--Rob
-Original Message-
From:
Hi,
the limits from te zaptel.h:
#define ZT_MAX_SPANS128 /* Max, 128 spans */
#define ZT_MAX_CHANNELS 1024/* Max, 1024 channels */
On Wed, 2005-05-11 at 10:45 -0500, Carlos Chavez wrote:
On Wed, 11 May 2005 07:31:17 +0300, Yousri Farouk wrote
Hello
Hi, does anyone know of * being used with icecast in any way. Does *
support vorbis at all? can anyone who is working on this give me a
shout.
Shidan
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi Peoples
I would be interested to hear from anyone who has
managed to get the Octtel SP4220 and asterisk talking together.
I am using the Octtel as a gateway for a PSTN line.
It passes the call on to Asterisk and then Asterisk moves the call to a
particular extension. Whilst I can
mine, on the stars of saturn
options:
Dione, Rhea, Titan, Mimas, Enceladus, Tethys, Hyperion, Iapetus, and Phoebe
Abhishek
--
Drishti-Soft Solutions Pvt Ltd
http://www.drishti-soft.com
On 5/12/05, Christopher Stephens [EMAIL PROTECTED] wrote:
Mine is called 'blacksun', as that's where it's
[EMAIL PROTECTED] wrote:
I've looked into this. The important reasons as to 'why this shouldn't
happen' are:
Requires a Database - (bad for flash, also very large)
Needs apache + php (+30 odd mb)
A fair whack of perl modules (+10mb)
== Too large, too cumbersome.
You can have
Hi Guys,
I actually had the idea in mind that the database would be located off
site... (not on the actual machine).
Still, with a larger flash card, would this not be possible ? (lol -
getting a larger flash card is not going to be an issue)
Just an thought.
Callum
Rob Thomas wrote:
Senad, the specs on the site for the minimum version seem to indicate a
HDD of at least 2Gb.
Am I wrong here... is there something that I am not seeing ?
Also... PBXware costs money and I don't want my cheap $1,000 unit to
become a $2,000 unit.
Any info would be appreciated.
Callum
Senad
Guys.
I just downloaded the recent firmware for GS GXP 2000 and I must say the
phone works great but... How do you make the conf button work?? Anybody
done that?
Also, with great dissapointment I must ask, where is ILBC support? GS web
page mentions it and the manual says it supports it
In article [EMAIL PROTECTED],
Daniel Salama [EMAIL PROTECTED] wrote:
I would really hate to having to install a digium card just for the
timer source.
[...]
I'd rather stay away from building custom kernels.
Any other suggestions?
No. If you don't have UHCI USB, those are the only two
hello,
i am french student and i want configure a Asterisk server.
when I want launch the server with the command safe_asterisk -vcf
the server answer : Asterisk ended with exit status 1
Asterisk died with code 1
what is the signification of it please ?
Are there any errors from /var/log/asterisk/messages? Or /var/log/messages?
Can you give some output from the startup when you execute asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tutu Lord
Sent: Thursday, May 12, 2005 12:58 AM
To:
Hi,
In regards to the previous thread about static and snapping on incoming
calls to the TE410P card when using a Dell 1850 server I now seem to be
getting significantly better call quality with two E100P cards. So far I
haven't been able to make any calls with detectable static on the line.
If I can help in beta testing or anything, please let me know Matt.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of mattf
|Sent: Miércoles, 11 de Mayo de 2005 10:15 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE:
Hi,I used C3640, but It was changed,
because of few DSP in it. However, configuration is same. It also depends on
used IOS version. Here are fragments from
configurations:AS5300:!clock timezone GMT
0
; in some Docs = necessary!isdn switch-type
primary-net5 ; I`m in Europe :-)isdn
I have download Astwind 0.1.1, my config is without Zaptel card and is mde
up of one computer without client which is connected on
my extensions.conf is :
[general]
static=yes
writeprotect=no
[globals]
[echotest]
exten = 600,1,Wait(3)
exten = 600,2,Echo
[local]
ignorepat = 9
include = echotest
Callum McGillivray wrote:
Senad, the specs on the site for the minimum version seem to indicate
a HDD of at least 2Gb.
Am I wrong here... is there something that I am not seeing ?
That was a type error... corrected. thanks :)
Also... PBXware costs money and I don't want my cheap $1,000
Run asterisk -vcf for test purpose.
Safe_asterisk is script which runs asterisk - so you wont get any messages on
screen
Br,
dmitry
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Tutu Lord
Sendt: 12 May 2005 09:58
Til:
Since we have SIP and ZAP servers, we name them something completely
original:
SIP01
ZAP01
ZAP02
TEST01
Whoa!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abhishek
Tiwari
Sent: Thursday, May 12, 2005 12:07 AM
To: Asterisk Users Mailing List -
Adam Goryachev wrote:
On Thu, 2005-05-12 at 07:35 +0800, Ronald Wiplinger wrote:
I have two ways to go to the voicemail box, either by dialing 8500
from the phone which received the voicemail (without a password) or
from another phone by dialing 8501 and key in the mailbox and the
Is there a way to make an outside call hear The
person at phone number is unavail, but when an internal extension
calls another extension, they hear The person at extension number
is unavail? I swear Ive
read this somewhere before but Im not typing in the right search. I probably
Chris Coulthurst wrote:
Is there a way to make an outside call hear The person at phone
number is unavail, but when an internal extension calls another
extension, they hear The person at extension number is unavail?
I swear Ive read this somewhere before but Im not typing in the
I have this warning popping up on one particular server.
chan_sip.c handle_response: Forbidden - Wrong password on
authentication for Notify.
I have looked around but cannot find what would be the cause of the warning?
Can anyone throw some light on this warning, why it is caused?
Thanks.
I know you can use slimserver as a music source, and slimserver supports
tons of formats, so maybe that's your answer.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shidan
Maybe not the place for this but thought
I'd post the info for others. I purchased a cisco 7960 off ebay and
needed to convert to SIP for *. I know * supports SCCP but I wont
go into that here. I'd read on voip-info.org that a contract could
be purchased for approx $8 to allow me to download the
I have this warning popping up on one particular server.
chan_sip.c handle_response: Forbidden - Wrong password on
authentication for Notify.
I have looked around but cannot find what would be the cause of the warning=
?
Can anyone throw some light on this warning, why it is caused?
Thanks.
Those files I indicated to check :
/var/lib/pgsql/data (on a redhat flavor)
pg_hba.conf - This one needs lines similar to
local all all password
host all all0.0.0.0 0.0.0.0 password
(not you probably want a more restrictive ip range /
yep, I think you're right that the voicemail.conf file is being
dynamically rebuilt. The reason that was not being reflected before is
that I had the voicemail.conf file open and therefore asterisk could not
write to it. However, I noticed that when I closed it and re-opened it,
that the changes
Colin
Similar to Gary's response in that I haven't seem many of these issues.
One that is similar, is that of you saying you need to press voicemail
key twice to get *97 (or eqivilent code)
This as I understand it is not a fault of snom, but a feature of
asterisk and the whole MWI protocol.
Thx for the pointer Peter.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Peter Bowyer
|Sent: Jueves, 12 de Mayo de 2005 12:45 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Grandstream GXP2000
I've got an 800 number through livevoip and have not noticed any failures
(incoming or outgoing). There certainly could have been a failure once in
a while, just have not heard or observed it. Of the several itsp's I've
tried over the last six to twelve months, its been the most stable and
Hello there,
I have configured my asterisk to run on
Mysql backend. But the Asterisk was unable to pick the peer details from the
database. This is how I configured the Asterisk to run with mysql on the
backend.
Edit /usr/src/asterisk/channels/Makefile,
change it to enable the
Yep it's called Jfax but it's a commercial service that there is a
charge for.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Coulthurst
Sent: Wednesday, 11 May 2005 11:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial
This has been great !! Thx Barney
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
barneySent: Jueves, 12 de Mayo de 2005 03:30 a.m.To:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600
On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote:
On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote:
For an internal historical reason all ours come from the legends of
Robin Hood. I used to work with a bunch of Lord of the Rings readers
and all the machine names came from
Hi everybody,
We are thinking in connect out PBX (with a new PRI card) to * (with
card TE110P) thought an E1.
We will have to configure the framing, coding, channels, etc...our doubt
is:
How must we select the signalling in * 'pri_cpe' or
'pri_net'? It's depend if our PBX card emulate to
Matt Riddell wrote:
For all italian speaking users please visit and contribute
to www.asterisk-italia.it!
I don't seem to be able to resolve that link.
Sorry! Not a very good start :-)
we had some dns propagation issues that are now solved.
The website is online now...
Thanks!
Hello All * users.
I have been looking for a way to allow GSM termination through Asterisk
to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on
asterisk with the ZAP channels via the Digium TDM 400P. I am unable to
find any place that can tell me the cost of the VoiceBlue with a
On Tue, May 10, 2005 at 12:01:17PM +0530, Sudhananda wrote:
I am running asterisk on one linux PC and want to talk through this server
using Kphone installed on 2 different PC's. These are the extra lines added
to sip.conf and extensions.conf respectively.
sip.conf
[jitha]
hello
i am using a call file. i want to insert delay before
execution of this call file. any idea how to do this
Channel: SIP/2000
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: default
Extension: 6000
Priority: 1
i am making a callback system.
when person rings to callback number this call
I'm not finding that on the Jfax website. Can you point me to more info on
how the act as a VOIP Fax Proxy?
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dean Collins
Etienne,
I am not sure I understand all what you require. Do you need to know the cost of
the voiceblue of 2N or you need to find solution that can allow you send GSM
calls ?
There are several alternatives:
1-) Voiceblue as you mentioned;
2-) You can buy a voip2GSM Gateway. To which you no
When the phones register with asterisk
Saved useragent Sipura/SPA841-3.1.2(d) for peer
I can see the firmware, which is handy for ensuring they are all up to date.
How can I list all the useragents?
Chris Mason
___
Asterisk-Users mailing list
With the recent service outage at Broadvoice, there has been a lot of
discussion here, on broadband reports, Voxilla, etc., regarding whether
VOIP is mature, or ready for the masses, etc.
One particular point I've seen repeated, and with which I agree:
we're willing to deal with less
David John Walsh [EMAIL PROTECTED] wrote:
I quite like the idea that came about earlier with regards to Romand
and Greek gods, I am thinking (if I ever get off the phone to google
today) of findind the roman and greek gods of communication..
You are thinking of Mercury and Hermes, the
it was a wheel. still went on it again an hour later once they put
it back on!!!
/never/ trust french theme parks :)
As a consultant focusing primarily on network performance and security
for the past twelve years, and working with clients in 40+ US States,
we've seen
- systems
I've some code to manipulate incoming Caller-ID - so its suitable for
replying to...
[sipdef]
exten = s,1,NoOp(FWD SIP: ${CALLERIDNAME} ${CALLERIDNUM})
; Alter incoming calles from pulver - add a '87'
exten = s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4)
exten =
- Original Message -
From:
Michael
George
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, May 12, 2005 5:22
PM
Subject: Re: [Asterisk-Users]
Kphone--asterisk--Kphone
On Tue, May 10, 2005 at 12:01:17PM +0530, Sudhananda
Can we get this guy booted off the list somehow?
-Original Message-
From: [EMAIL PROTECTED] [mailto:MAILER-
[EMAIL PROTECTED]
Sent: Thursday, 12 May 2005 8:36 AM
To: Dean Collins
Subject: failure notice
Hi. This is the qmail-send program at smtp.register.it.
I'm afraid I
Same thing happend to me. I order a 954-XXX-XXX DID on 04-06-2005 and
I'm still waiting. My order status also says pending.
On 5/11/05, BJ Weschke [EMAIL PROTECTED] wrote:
I ordered a 973-XXX- and 585-XXX- DID from them on 2/3 and 2/7
of this year respectively.
Their customer service
Could someone please comment on the current state of chan_capi,
chan_misdn and chan_modem channel drivers in terms of functionality and
stability. Specifically, which channel driver would be best for a
passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that
chan_capi distinguishes
I solved the problem by rechecking my configuration files, namely mgcp.conf and
extensions.conf.
I changed the EPIDx strings in the ATA188 to a001 and a002 (and changed
accordingly in other config
files), the context from default to ext_mgcp in mgcp.conf and set all the ports
to 2427 and now
2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you
just register the voip2GSM devise to Asterisk and then it is ready to receive
and send calls just like any other sip phone. Cost of this is around 400 USD /
UNIT
That is interesting. What is the make and the model
Also off the top of my head.. How about:
specify a context in voicemail.conf:
[outward-dial-by-name]
2125551212 = 1000,John Smith
301212 = 1000,George Lucas
or if you use 9 to dial out:
[outward-dial-by-name]
92125551212 = 1000,John Smith
9301212 = 1000,George Lucas
Again, I have not
Has anyone heard of a working Open Source Softphone
compatible with the MGCP protocol ?
Right now, I know of the eyeP softphone, but it is not Open
Source.
Thanks for any help.
JFD
___
Asterisk-Users mailing list
On May 11, 2005 05:15 pm, Armin Lediger wrote:
I am trying to install asterisk 1.0.7 on a VIA EPIA 5000 board - anyone
of you already managed to do so? I got V1.0.6 running, but 1.0.7 seems
not to compile.
Just a correction; this isn't about a VIA chipset; this is about a VIA
processor.
it seems you are right, you need the context there, but than I cannot
use it in Realtime anymore, since I have more than one context
Why would having more than one context stop you fomr using RealTime?
Doesn't stop us.
I would than need for each context an extra extension number.
It
Hello,
Being totally fed up with the lack of quality and reliability from both
VoicePulse and BroadVoice,
We are switching to a direct IP connection to Global Crossing. We've
installed a local point-to-point T1 into their CO, and they will
give/take SIP g729a directly and act as the gateway for
I also get this when doing a Manager Click2Dial application except the ^^ in
the error go on a few thousand times. The call still completes but you still
get the error.
-Matthew
Mark Elkins wrote:
I've some code to manipulate incoming Caller-ID - so its suitable for
replying to...
[sipdef]
I don't think my first posting went thru.
I am trying to set up Asterisk for the first time. I am new to this.
I am using [EMAIL PROTECTED]
I have a TDM400P with one FXO and one FXS
The system is working for outgoing calls and if I test incoming calls using
.
But when doing an actual call
This is what I getafter Zap/4-1 answer I can press # and
the call go thru just fine..I just can find a way to force the # go in
automaticly @ end... :-( any ideas?
===Connected to
Asterisk
Hi all,
When I enabled faststart in oh323.conf, calls from H323 endpoint to
SIP phones could not complete. The originating phone kept ringing when
calls were answered by SIP phones.
fastStart=yes
h245Tunnelling =yes
h245inSetup=yes
Please can you advise.
Many Thanks.
You should put your asterisk into verbose mode using asterisk -c
or if you are using a server asterisk -r and you can trace out
what happens and it will be in the log file called full in the
/var/log/asterisk directory and then you can probably figure out what
happened. Your incoming call
trixter http://www.0xdecafbad.com wrote:
They paid 100% of the *UNDISPUTED* charges but nothing is said about the
disputed ones. Typo or intentional? It also sounds to me like its an
access charge issue, but I may be reading too much into this.
Sounds like BroadVoice paid their bill
It sounds like you don't have USB support compiled in the kernel.
Chris
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 11:55 PM
Subject: Re:
Could someone please comment on the current state of chan_capi,
chan_misdn and chan_modem channel drivers in terms of functionality and
stability. Specifically, which channel driver would be best for a
passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that
chan_capi distinguishes
On Thu, 2005-05-12 at 17:31 +0400, Jean-Michel Hiver wrote:
2-) You can buy a voip2GSM Gateway. To which you no longer need hardware, you
just register the voip2GSM devise to Asterisk and then it is ready to receive
and send calls just like any other sip phone. Cost of this is around 400 USD
I believe *ANI*DNIS
That's how Asterisk sends it when I set my t1 line to featd.
In /etc/asterisk/zapata.conf
signalling=featd
not much to go on, but a little!
-Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, May 12,
Thanks I will give that a try.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Thursday, May 12, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Incoming calls picked-up then simply
On Thu, 2005-05-12 at 09:08 -0500, Dan Perik wrote:
trixter http://www.0xdecafbad.com wrote:
They paid 100% of the *UNDISPUTED* charges but nothing is said about the
disputed ones. Typo or intentional? It also sounds to me like its an
access charge issue, but I may be reading too much into
Hello
Sorry for english speaking peaple, but I just help this beginner in our
natural language : French ;-)
Je suis Français aussi, si tu as besoin d'un peu d'aide tu peux me joindre
directement par mail
Pour tester ta config : asterisk -gc
Bonne chance
I've found on wiki that there is a variable called ${BLINDTRANSFER}
which should contain the channel (or a number) of user that made a blind
transfer of
a call to another extension.
Also I've found a patch for chan_sip to add support for ${BLINDTRANSFER},
but it's not working at all (chan_sip
What I have discovered is that my motherboard only supports usb-ohci
and not usb-uhci. Reading on the wiki, it says that ztdummy requires
usb-uhci.
To make things worse, I slapped in a TDM22B just to get timer
support, only to discover that the machine kept crashing because of a
hardware
Hi,
That is interesting. What is the make and the model that you are
referring to? Is there a website with more info?
As for the models, we sell them as OEM. You may contact me offlist if
interested. Better priced and more powerful than existing devises out there.
I currently use
-Original Message-
snip
I just downloaded the recent firmware for GS GXP 2000 and I must say the
phone works great but... How do you make the conf button work?? Anybody
done that?
---I just put in a system with 25 of these and have the same issue. Looks
like the conf button will
I am posting this in case someone need
help.
=
YOU THA
MAN!
No sure how I will repay you, but
anything you need, just let me know!
Thank you, thank you,
Hi
I've a problem with a gnugkv2.0.7
I've compiled gnugk successfully
I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully
When i run gnugk i have this error:
error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot
open shared object file No such file or
The good thing about gsm files and the fact that they are
headerless is that you can simply cat files together. You just need to find the
right sound files to do so.
Then program your
dialplan to play the message before sending the person to voicemail. I would
zero out the unavailable and
Can anyone provide more information on switch or point me to where I
can find more about it?
The only I've been able to find on the wiki is:
http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers
and towards the bottom of (section Forwarding to another Asterisk):
This is what I got:
May 12 11:12:53 VERBOSE[1376]: -- Starting simple switch on 'Zap/4-1'
(Note that the line went dead on the calling phone before this next stuff
ever appeared)
May 12 11:13:01 WARNING[1376]: CallerID returned with error on channel
'Zap/4-1'
May 12 11:13:01 VERBOSE[1376]: --
Hi Folks,
I am planning to make a little project of voice recognition.
I already browsed Voip Wiki and found some solutions.
Before putting my hands on it to just do a little demo menu,
I would like to hear from the list any succesful case using voice
recognition and Asterisk.
Best Regards,
Folks!
I am looking at a couple of models of Fixed GSM Gateways for the Purpose of
VOIP connectivity and specifically to work with Asterisk. I found that these
can be imported into USA for about $99.99 or about that. This is a one channel
unit just like tellular, one of them has GPRS.
Take a look at the weight option in queues.conf. Available in CVS
only I believe.
Callum McGillivray wrote:
Hi All,
We have been playing around with call queues and asterisk and now have
everything set up the way that we want it, bar 1 thing.
When we have a scenario of an agent logged into
Hello All,
Can anyone u pls tell me the context pattern i need to add on sip.conf
and extension.conf for incoming calls ... the senerio is i have a
provider who routes a UK DID to my IP previously i was using
ATA186 and calls were coming on ATA186 via sip and phone was connected
to port 1 ..
huh? That's a TDM/RBS type question.
I've not seen most implementations of SIP interconnections doing
things like that?
On 5/12/05, Adam Robins [EMAIL PROTECTED] wrote:
Hello,
Being totally fed up with the lack of quality and reliability from both
VoicePulse and BroadVoice,
We are
I have some beefy dual-Xeon servers that I will be using for Asterisk
VoIP applications (i.e. no Zaptel cards). Using 2.6.11-1.14_FC3smp
as the kernel (Fedora Core 3), and currently with Asterisk STABLE.
My question is concerning the CPU setup, as I've seen conflicting or
out-of-date suggestions:
Sorry for my delayed response Selon ,
I am setting up a test Asterisk box in our company to replace our
current switchboard and well -
GSM connection was one of the requirements for me to do to allow
asterisk to replace our switchboard.
(The others are not going to well... or they are finally
Good day all
I installed asterisk-addons and now its logging nicely in my database
But I want it to log in my usual log csv as well
Please Let me know
Thanks
Altus
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Thu, 2005-05-12 at 11:27 -0400, Kanuri, Seshu (Company IT) wrote:
Folks!
I am looking at a couple of models of Fixed GSM Gateways for the Purpose of
VOIP connectivity and specifically to work with Asterisk. I found that these
can be imported into USA for about $99.99 or about that.
Bharat M. Sarvan wrote:
Hello Everybody,
I am having problems with starting Asterisk.
The message what I am getting is;
May 11 15:41:32 WARNING[5031]: res_musiconhold.c:728 moh_scan_files:
Cannot open [cdr_addon_mysql.so]May 11 15:41:32 WARNING[5031]:
Hi Isamar,
There is a trial project underway for Asterisk and www.tellme.com but
this is a commercial implementation of Speech Recognition using external
resources and infrastructure.
This will not be free.
Let me know if you have a commercial application that has funding behind
it.
Regards,
Are you sure you have context=from-pstn in your zapata.conf for the
fxo channels?
Julian.
On 5/12/05, fhunter [EMAIL PROTECTED] wrote:
I don't think my first posting went thru.
I am trying to set up Asterisk for the first time. I am new to this.
I am using [EMAIL PROTECTED]
I have a
Is anyone here able to make calls to FWD via IAX? I haven't beenable to
for some while. I'd like to get to the bottom of the problem. There's
been little response in the FWD support forum thus far.
I can call my own number and it rings my server, but I cannot call any
other number. It generates
Hi team,
Not long ago a bunch of us were posting reports of a strange phenomenon
where voice quality would pack up completely from time to time,
typically resulting in loud crackling on the line and/or the voice
channel breaking up completely. With our installation it would occur
from time to
This seems to be par for the course: You'll get a DID and poof, it's
gone! Nobody answers the phone and nobody responds to tickets.
For example:
http://www.sixtel.net/tickets/view.php?ticket=xojnikrapqofyaspej
-Steve
Wiley Siler wrote:
Anyone have an opinion about these guys and their recent
Hello All,
Can anyone u pls tell me the context pattern i need to add on sip.conf
and extension.conf for incoming calls ... the senerio is i have a
provider who routes a UK DID to my IP previously i was using
ATA186 and calls were coming on ATA186 via sip and phone was connected
to port 1 ..
Interactive Intelligence has a commercial Speech recognition API for
this purpose.
Check http://www.inin.com
Or the specific Vocalite engine page at:
http://www.inin.com/Products/vocalite/vocalite.asp
Seshu Kanuri
NOTICE: If received
As far as I can see in my installation, it does both.
Nathan
Altus Snyman wrote:
Good day all
I installed asterisk-addons and now its logging nicely in my database
But I want it to log in my usual log csv as well
--
-
Nathan E. Pralle
Give the director a serpent
Hi Tony, check out my recent post regarding our experiences with
Hyperthreading and * with Zaptel cards.
We have a few machines in the wild that *do* run Hyperthreading but no
Zaptel cards and these work absolutely fine. My understanding is that
the Hyperthreading problems are purely related
Odd problem here--I just got a couple of Cisco 7960s from Ebay that are not
functioning as expected..
These 7960s can't seem to be unlocked for manual configuration via any
mechanism that I can find. If you go to settings, there is no option 9
(unlock). Available options stop at 4
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