Hello,
We are glad to announce new release of our advanced billing and routing
package for Asterisk - MOR v0.7
It is complete solution for VoIP billing and routing for advanced and
start-up telecoms, carriers, voip calling card operators and ISPs.
Demo available online, as LiveCD or as
I recommand what I know and I sell. Vierling Ecotel VoIP for small
capacities and Vierling VTM-pro for ISDN interfaces
Regards,
Hakem,
2008/11/29 Michael Graves [EMAIL PROTECTED]
Portech makes larger rack mounted modular multi-channel gateways as
well. Not sure about the ISDN interface, but
I'm about to begin working on an ivr project to do database backed
scheduling. I would like to use text to speech in some places. What are
the differences in using festival vs. Cepstral? How are they similar, how
are they different? Is one really better than the other? How and Why?
Thanks,
On Tue, Dec 02, 2008 at 11:17:11AM +0100, Olivier wrote:
2008/12/2 Tzafrir Cohen [EMAIL PROTECTED]
On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote:
Hi,
Testing latest 1.6.1, it occurred to me I had to add a couple of noload
statements in /etc/asterisk/modules.conf to
Hello All,
I currently have an Asterisk Box, running a callcenter with 04 queues. I set
queues.conf with persistentmembers=yes in the general section as follows:
[general]
monitor-type = MixMonitor
persistentmembers = yes
However when I perform any kind of restart in the Asterisk
Dear Sir,
My Asterisk server is sending periodically the below SIP packets
Retransmitting #4 (NAT) to 68.62.168.138:5060:
OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK37f337ed;rport
From: asterisk sip:[EMAIL PROTECTED];tag=as078bf319
To: sip:[EMAIL
Doug wrote:
At 18:56 12/1/2008, Tilghman Lesher wrote:
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed to download public domain
material? That kind of restricts
Any idea? Please I need advice.
Thanks!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian
Sent: lunes, 01 de diciembre de 2008 11:58 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Parking calls
Hi,
How can I park
On December 1, 2008 07:21:33 pm Doug wrote:
Hmmm. When our users are pounding the network
with BitTorrent traffic, we just shut them down
and wait for them to complain. It's against our
Acceptable Use Policy, and causes all sorts of
VOIP headaches.
As someone who is the technical lead for
Hi,
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party answers).
It seems to me that asterisk keeps looking for this macro in extensions.conf
and not in extensions.ael.
I tried both (and variations of those with ^
Doug [EMAIL PROTECTED] writes:
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
Yes, just like the airlines need to somehow control those users who
keep showing up to
Hi,
I think this have been talked over several times but I couldn't find any
answer.
Sorry for asking.
I want from dialplan, to transfer a callee to a context-extension-priority
that would play a given fax file to callee (callee is supposed to be a fax
number).
I can get caller's channel id
Olivier schrieb:
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party answers).
It seems to me that asterisk keeps looking for this macro in extensions.conf
and not in extensions.ael.
I tried both (and
Err... to follow-up just regarding error messages:
On Tue, Dec 02, 2008 at 11:17:11AM +0100, Olivier wrote:
2008/12/2 Tzafrir Cohen [EMAIL PROTECTED]
On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote:
Hi,
Testing latest 1.6.1, it occurred to me I had to add a couple of noload
hakem Ta schrieb:
I recommand what I know and I sell. Vierling Ecotel VoIP for small
capacities and Vierling VTM-pro for ISDN interfaces
When I gave the Vierling Ecotel SIP-GSM gateway a try (years
ago) it was a nightmare.
There was an IP address printed on a sticker on the device -
didn't
Barton Fisher wrote:
any ideas?
None so far, what version of Asterisk?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and
Which non-english language do you have in mind ?
Both should differ on this.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hello there...
Noticed some strangeness going on with mixmonitor and chanspy, the called
(External SIP) party seem to be responding before the calling party
(Internal SIP) on call recordings and also when you listen in using chanspy.
as far as the agent (calling party) is conserned the
Ronald Wiplinger (Lists) wrote:
I know I can setup asterisk without Internet at all and it works as
local pbx.
Would an asterisk box work with a dynamic IP, with a dyndns name?
What must I take care if I try that?
I had my * server behind my adsl router that was getting a dynamic Ip
On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy [EMAIL PROTECTED] wrote:
Everyone--
I've just made some major changes to the CDRfix2.rfc.txt
file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
to accommodate the Leg approach instead of a
channel-based approach.
Hi murf,
I've got a
2008/12/2 Tzafrir Cohen [EMAIL PROTECTED]
On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote:
Hi,
Testing latest 1.6.1, it occurred to me I had to add a couple of noload
statements in /etc/asterisk/modules.conf to remove ERROR messages, when
starting Asterisk.
(I don't imply
Hello,
Asterisk 1.4.22 keeps crashing on Solaris 5.10 i386.
ast_dynamic_str_thread_build_va() seems to be passed some kind of
garbage (see attached dbx output) which ultimately brings down the
whole process. As a workaround, I've set the debug level to 0 for now.
Should I submit this as a bug?
For a ptmp setup where you have multiple phones.
Or even a single phone if the port is set to ptmp.
Proof of this point is the way I am using our B410P card. Ports 1 and 2
are TE (ptp) and ports 3 4 are NT (ptmp).
I have a single ISDN modem connected to port 3 and the B410P would not
even
On Tue, Dec 02, 2008 at 10:02:06AM -, Andrew Thomas wrote:
asterisk-users@lists.digium.com has now been added to the filters white
list!
Anyway, 100ohm termination isn't required for ptp - but is required for
ptmp.
For a ptmp setup where you have multiple phones.
--
asterisk-users@lists.digium.com has now been added to the filters white
list!
Anyway, 100ohm termination isn't required for ptp - but is required for
ptmp.
I know the DAHDI package(s) no longer include make b410p - hence the
reason it is included in the docs.
Philipp Kempgen schrieb:
Olivier schrieb:
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party answers).
It seems to me that asterisk keeps looking for this macro in extensions.conf
and not in extensions.ael.
On Tue, Dec 02, 2008 at 08:30:34AM +0100, Olivier wrote:
Hi,
Testing latest 1.6.1, it occurred to me I had to add a couple of noload
statements in /etc/asterisk/modules.conf to remove ERROR messages, when
starting Asterisk.
(I don't imply those ERROR messages were fatal to Asterisk but as a
Hi,
1. Has anyone got any success when send a TIFF file form one zoiper
softphone to another ?
I tried using Zoiper 2.18 free edition in windows but I'm seeing 415
Unsupported media replies.
2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read :
Also, try using:
2008/12/2 Philipp Kempgen [EMAIL PROTECTED]
Philipp Kempgen schrieb:
Olivier schrieb:
How can you use Dial application M(x) option from extensions.ael ?
(As a reminder, this M(x) executes macro x when Dial called party
answers).
It seems to me that asterisk keeps looking for this
Olivier wrote:
As latest asterisk-libpri-dahdi is introducing dahdi support of B410P,
can we use High Performance Echo Canceling addon with B410P ?*
Yes, DAHDI echo cancellers work with any DAHDI supported interface.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The
On Tuesday 02 December 2008 01:21:46 Giedrius Augys wrote:
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
WHERE
In my experience cepstral has always had much nicer sounding voices, but I
haven't tinkered too much with either. There is a reason one is pay and one
free though J I believe cepstral is still offering demo's, I'd download each
and see which one gives you the performance you're looking for.
Festival is a free voice that sounds like a machine. Cepstral is a fee
based human voice ($30 USD per voice per CPU). They are similar in that
they both produce mechanically timed output. IMO, you should use festival
if this isn't a customer based interface. If it is a CBI, use cepstral and
if
Hi All,
I need to stop the transfer feature on particular sip user.
I am using linksys phone and it has set the forwarding enable to another
user.
I have three users 2101, 2102, 2103.
2102 is registered in linksys phone with forwarding enable to 2103.
But is there any procedure in asterisk that we
On Mon, Dec 1, 2008 at 7:57 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
JD schrieb:
As to the idea of piping to a deamon via socket or dbus: how would
asterisk behave if the daemon froze or worse, it lagged?
I have implemented something similar with the Dial command. We had a
customer that
Festival sucks. Cepstral sucks less. The End.
In my experience, it depends on the specific app, who's paying, and who's going
to be the victim, err...user listening to it. This is the difference between
domain/context specific phrases/words to pronounce vs. general stuff, a client
on a tight
Using Asterisk 1.4.21.2 I am seeing pairs of warning logs of the form:
asterisk[1432]: WARNING[1432]: chan_sip.c:11629 in
build_reply_digest: use realm [x] from peer [x][x]
These occur once an hour and the x matches the account name for my
ITSP. My sip.conf setup for this
Give this a go:
exten = s,n,MYSQL(Query resultid ${connid} SELECT `name` FROM `cnam`
WHERE `ani` = '${CALLERID(number)}')
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort
Sent: Tuesday, December 02, 2008 3:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] cepstral vs festival
I'm about to begin working on an ivr project to do database backed
On Dec 2, 2008, at 7:01 AM, Grey Man wrote:
On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy [EMAIL PROTECTED] wrote:
Everyone--
I've just made some major changes to the CDRfix2.rfc.txt
file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
to accommodate the Leg approach instead of a
Hi,
Is there a way to page a Polycom phone that is already in use (if, of
course, the call isn't on speakerphone already)?
Mike
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
Mike wrote:
Hi,
Is there a way to page a Polycom phone that is already in use (if, of
course, the call isn't on speakerphone already)?
I've never been able to find a way. Any attempt I made either put the
existing call on hold to auto-answer the page or the page just rang at
the
2008/12/2 Tilghman Lesher [EMAIL PROTECTED]
On Tuesday 02 December 2008 01:21:46 Giedrius Augys wrote:
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT
You can send an IM to the phone with a text message. Assuming that the
phone has more than 1 line and at least one is open, the call should go
through without effecting the existing call. To do this from the dialplan,
you could set up something like this:
Exten = 411,1,Dial(SIP/100,1)
Exten =
Hi,
i am running Asterisk 1.6.0-beta4 and i have some trouble with the
Bridge-Application.
Here is what i want to do:
1) Caller A calls an extension and is connected to an AGI-Script.
2) Doing stuff and originating a second call per Manager Interface
3) Call will be set to an extension with
Right after sending the email, the solution came to me. I have fooled
myself: A ManagerEventListener kicked in an issued an HangUp Action on
the second channel right after the Bridge ...
The Bridging workes perfectly after fixing the EventListener.
Have a nice day, i will go home and hit
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
manager.c:1732: error: (Each undeclared identifier is reported only once
manager.c:1732:
On Tue, Dec 2, 2008 at 8:22 PM, Dave Fullerton
[EMAIL PROTECTED] wrote:
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
manager.c:1732:
On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
manager.c:1732:
Tzafrir Cohen wrote:
On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
It bombs out when compiling manager.c
On what platform is it?
Fails on CentOS 5x86 as well.
jlc
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Tue, Dec 2, 2008 at 9:56 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error:
hi
this is mi first email and just for say hello.
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Geraint Lee wrote:
Hello there...
Noticed some strangeness going on with mixmonitor and chanspy, the
called (External SIP) party seem to be responding before the calling
party (Internal SIP) on call recordings and also when you listen in
using chanspy. as far as the agent (calling party)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Atis Lezdins
Sent: Tuesday, December 02, 2008 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2,
At 04:03 12/2/2008, Benny Amorsen wrote:
Doug [EMAIL PROTECTED] writes:
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
Yes, just like the airlines need to
Hi,
I appear to have fixed this issue. If I turn off call recording in queues.conf,
the bridging succeeds. I did this by commenting monitor-format and
monitor-type.
thanks,
Tony Gaspar
From: Tony Gaspar [EMAIL PROTECTED]
To:
At 07:00 12/2/2008, SIP wrote:
Doug wrote:
At 18:56 12/1/2008, Tilghman Lesher wrote:
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed to download public
Doug wrote:
At 04:03 12/2/2008, Benny Amorsen wrote:
Doug [EMAIL PROTECTED] writes:
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
Yes, just like the
At 07:57 12/2/2008, Andrew Kohlsmith (lists) wrote:
On December 1, 2008 07:21:33 pm Doug wrote:
Hmmm. When our users are pounding the network
with BitTorrent traffic, we just shut them down
and wait for them to complain. It's against our
Acceptable Use Policy, and causes all sorts of
Doug wrote:
At 07:00 12/2/2008, SIP wrote:
Doug wrote:
At 18:56 12/1/2008, Tilghman Lesher wrote:
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed to
It is not a parking solution.
Sebastian wrote:
Any idea? Please I need advice.
Thanks!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian
Sent: lunes, 01 de diciembre de 2008 11:58 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
This seems to be an AGI/Music on Hold solution to me. For parking to work,
you would have to know which lot you parked the call in and pick it back up
when done, assuming that another user did not pick it up and that the caller
did not hang up.
From the dialplan, you would call an AGI. The AGI
On Tuesday 02 December 2008 12:22:16 Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
I'll get a new release candidate out either this afternoon or tomorrow;
I'm currently working on ensuring that 1.6.0.3 will not be a regression from
1.4.23.
--
Tilghman
Hi,
I need to run ztdummy for Paging, but now that this is all become dahdi I
don`t really know where to start. I did build dahdi before building
asterisk, but that`s it.
I find it hard to find any documentation referring to dadhi instead of
zaptel.
I have no Digium hardware, but I
Mike wrote:
Hi,
I need to run ztdummy for Paging, but now that this is all become dahdi
I don`t really know where to start. I did build dahdi before building
asterisk, but that`s it.
I find it hard to find any documentation referring to dadhi instead of
zaptel.
I
hi
i need an open source callcenter manager system like queuemetrics but
opensource any one know any?
i prefer to search before start a new one
thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
I guess my question is more basic than that: I have two brand new 1.4.22
systems, one with DAHDI apparently running well (dahdi start looks god)
running well with Paging, and the other with FATAL errors modules cannot be
found and paging not working.
I seem to remember installing both Asterisks
there is any card?
you musnt load any module that is not going to be used.
you can get some errores if the card is misconfigured
trai dahdi_cfg -vvv you will get some idea of the problem
David
2008/12/2 Mike [EMAIL PROTECTED]
I guess my question is more basic than that: I have two brand new
On Dec 2, 2008, at 9:41 AM, Erik (Caneris) wrote:
Festival sucks. Cepstral sucks less. The End.
In my experience, it depends on the specific app, who's paying, and
who's going to be the victim, err...user listening to it. This is
the difference between domain/context specific
Although QM is not open-source, it is extremely affordable and high
quality for the price.
David fire wrote:
hi
i need an open source callcenter manager system like queuemetrics but
opensource any one know any?
i prefer to search before start a new one
thanks
David
--
(\__/)
thanks for your answer but i need an opensource
i know quemetrics is good but i need an open source.
thanks
David
2008/12/2 Alex Balashov [EMAIL PROTECTED]
Although QM is not open-source, it is extremely affordable and high
quality for the price.
David fire wrote:
hi
i need an open
I did build dahdi before building asterisk, but that`s it.
No problem. But what steps did you use? Did you edit *any* dahdi related
configs? See the voip-info url below.
I find it hard to find any documentation referring to dadhi instead of zaptel.
:) Yeah, it's not the most documented aspect
John Todd wrote:
Erik -
Have you found RealSpeak to be worth the cost? Can Cepstral, with
the hourly $ spent on tuning, be made to be a reasonable substitute?
It's been a while since I did a head-to-head comparison between
Cepstral and (anything else) so I did a quick demo of the
Do you need free or do you need opensource?
What is your budget? Number of agent seats?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
On Tue, Dec 2, 2008 at 6:18 PM, David fire [EMAIL PROTECTED] wrote:
thanks for your answer but i need an
line 0: Unable to open master device '/dev/dahdi/ctl
Well that probably explains it, because there is no such file. But as I am
not a linux expert (comfortable linux user at best), I am not sur where to
go next.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__ Information from ESET Smart Security, version of virus signature
database 3659 (20081202) __
The message was checked by ESET Smart Security.
http://www.eset.com
Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned
in my other post, I don`t have a /dev/dadhi fileI don`t know why it
wasn`t created or where to go from here.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph L.
my budget is 0 rigth now
and i want opensource because i want to customice it... can program in
Java
PHP
C
C++
.NET (i am not proud of it)
so i want to customice it for my clients (furute clients i havent any now)
and give the improvements to the comunity.
Davif
2008/12/2 Steve Totaro [EMAIL
ok
dont pay attention to that file for now...
do you have any card on that machine? any digium card or any other brand?
or not?
if not the problem is that you dont need to load any module (just the
dummy one)
if you have any card you have a problem in the config.
David
2008/12/2 Mike [EMAIL
I have no cards (nothing dahdi related). Why is my other server, built with
default settings, working then?
Still
what do I do ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 02, 2008 19:00
To: Asterisk Users Mailing List -
Sorry, that worked, I just removed all modules from the modules file in
/etc/dahdi.
But that doesn`t explain why my other card-free PC is working perfectly with
the default modules files while this one isn`t
Thanks though, that saved my behind. But an explanation, if an easy one can
be
At 12:44 PM 12/2/2008, you wrote:
At 04:03 12/2/2008, Benny Amorsen wrote:
Doug [EMAIL PROTECTED] writes:
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
Have you looked at AMS?
http://www.intuitivecreations.com/contributions/AMS/
PaulH
David fire wrote:
my budget is 0 rigth now
and i want opensource because i want to customice it... can program in
Java
PHP
C
C++
.NET (i am not proud of it)
so i want to customice it for my clients
options are:
-you made a mistake
-only g'ds know
probably when you installed dahdi you made make config in only one pc.
David
2008/12/2 Mike [EMAIL PROTECTED]
Sorry, that worked, I just removed all modules from the modules file in
/etc/dahdi.
But that doesn`t explain why my other
Hello Philipp,
All what you says was true and I am first to witness it. Ecotel VoIP was a
new product at that time needed some time like wine to become good
1-) Files did not cleanup correctly ;
2-) Lot of erratic issues that we have also greyed hair with;
3-) Stability issues, ...etc;
When I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Todd a écrit :
My results: The RealSpeak sample was more clear than the Cepstral.
But by how much? I should probably test with more than just that one
phrase, but I can't say I'd prefer RealSpeak significantly over
Cepstral in this
David fire wrote:
hi
i need an open source callcenter manager system like queuemetrics but
opensource any one know any?
i prefer to search before start a new one
You'll be pushing to find something even close to QueueMetrics' quality
available in open source. The closest I'm aware of is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This has been an interesting discussion about cepstral. My question is
why it doesn't appear to be available for 1.6 yet? This thread has
piqued my interest in the product but a visit to Digium's website seems
to point to it being a product for
Jean-Denis Girard wrote:
The price of RealSpeak is not far from an order of magnitude higher
compared to Cepstral.
Only an order of magnitude? They've reduced it a lot then. :-)
Steve
___
-- Bandwidth and Colocation Provided by
Try just modprobing the module and see what happens. This worked for
me when it was zaptel.
on Tuesday 12/02/2008 Mike([EMAIL PROTECTED]) wrote
I have no cards (nothing dahdi related). Why is my other server, built with
default settings, working then?
Still what do I do ?
Erik -
Have you found RealSpeak to be worth the cost?
Actually my last note was probably a bit misleading because in the particular
cases I mentioned RealSpeak, the platform wasn't Asterisk and Cepstral wasn't
even on the radar.
Can Cepstral, with
the hourly $ spent on tuning, be made
2008/12/3 Steve Underwood [EMAIL PROTECTED]
Jean-Denis Girard wrote:
The price of RealSpeak is not far from an order of magnitude higher
compared to Cepstral.
Only an order of magnitude? They've reduced it a lot then. :-)
1 order of magnitude = x10
Then, shall we say 500$/simultaneous
2008/12/3 Joseph L. Casale [EMAIL PROTECTED]
Do an lsmod and look for something like so:
[EMAIL PROTECTED] ~]# lsmod | grep dahdi
dahdi_dummy38984 0
dahdi 231760 9
dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp
crc_ccitt
Hi
I had tried your sugesntion and added in the appropriate context but got
this error message
Rejected connect attempt from 192.168.254.185, request
'[EMAIL PROTECTED]' does not exist
any ideia?
thanks
coco wrote:
Hello
I asked the same thing some time ago, but nobody answered.
I
On Tue, Dec 02, 2008 at 06:47:02PM -0500, Mike wrote:
line 0: Unable to open master device '/dev/dahdi/ctl
Well that probably explains it, because there is no such file. But as I am
not a linux expert (comfortable linux user at best), I am not sur where to
go next.
This probably
On Wed, Dec 03, 2008 at 07:43:55AM +0100, Olivier wrote:
2. How can you check dahdi is running ?
cat /sys/module/dahdi/version
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com
2008/12/3 Tzafrir Cohen [EMAIL PROTECTED]
On Wed, Dec 03, 2008 at 07:43:55AM +0100, Olivier wrote:
2. How can you check dahdi is running ?
cat /sys/module/dahdi/version
Thanks !
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED]
98 matches
Mail list logo
