Sorry 4 a.m. I'm kind of tired and I slipped a password. :S
Already changed it. Sorry!
Juan Luis Moyano wrote:
> Hi, im trying to setup ASTCC but I'm getting it difficult. I've
> correctly set up the mysql database astcc and added a brand, trunk,
> route and a card as follows:
>
> brands
> +-
Hi,
It
seems that my flash panel only works when I specify my ip address and not the
host name.
I've
tried quite a few things (change host file, dns resolve, proxying….) but couldn’t
get it to work.
Anyone
knows how to solve this?
Thanks,
Ohad
___
Hi,
I'm pulling my hair down and getting bold :-) . I have Asterisk between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)
I'm trying to do just plain transfer of call from pbx to ISDN through
Asterisk...
It seems like PBX hangsup, when call is progressi
Ypek,
> I would like to know how can I manage to implement a table which translates
> an extension number into a phone number. Let see an example:
There are many ways of doing this. You could map the extensions to
phones in extensions.conf, via the internal database or via an
external database, o
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
+--+--+--+--+--++--+--+
| name | language | inc | publishednum | did | markup |
> I have been trying to open an account with voipreach.net for over
> a week now and I have not gotten any response from them as yet.
> None of their phone numbers are working.
They didn't respond to my emails either... Tixter is right, forget
about them if they don't even care to reply to take you
On Thu, 23 Jun 2005, Massimo De Nadal wrote:
> Have you planned to integrate some echo cancel feature ?
Echo cancelling (if the card supports it) is already implemented.
As far as I know the Eicon Diva Server cards are the only cards supporting
echo cancel via onboard DSPs.
Armin
> Armin Schindl
Does anyone have a MAX/APX with working ingress PRI calling name?
I recently acquired a MAX TNT on the cheap and it's integrating fine
except for one thing. In the 11.0.0 release notes, it is stated that
ISDN calling name will, if present and permitted by presentation
flags, be added to the From:
It's a Digium single-port job. No other timing sources aviailable (the * box IS
the pbx).
qrss wrote:
What kind of card are they using? Is there only 1 telco circuit?
If so, then I'm thinking their card should have detected the loss of
service and switched to it's internal clock. Do they hav
Have you planned to integrate some echo cancel feature ?
Armin Schindler ha scritto:
Hi all,
I would like to announce the first release of the chan_capi
channel driver on sourceforge.net
The package is available for download with name
chan_capi-cm-0.5
and is the current CVS HEAD.
It is d
On Wed, 2005-06-22 at 21:59 -0700, George Pajari wrote:
> Adam Robins asked:
>
> >Shouldn't [Asterisk] be smart enough to go to Zap/4 as the only available
> >port in the group [with a live trunk]?
> >
> >
> Adam Goryachev wrote:
>
> >No, asterisk doesn't do dialtone detection.
> >
> But this i
I am experiencing a very wired
problem.
Some of my cdr are lost.
I use logging cdr to csv, mysql and odbc. But some
of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is
related to asterisk functioning.
I am running asterisk 1.0.7; this is simple
configuration fil
Change the dialplan in your spa3k with something like:
(xx.|*x.|**x.)
This way you can dial any number, even starting with * or **
Martin Roy ha scritto:
How can I dial *67 on a Sipura 3000 if I dial from a SIP phone
connected on an asterisk server. I always get a message saying that
authe
I'm following the instruction from João Amaro from the
page
http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html
Everything went fine until I run the 'make' command
under asterisk-oh323-0.6.5. I got the error message
chan_oh323.c:5220: too many arguments to function
`ast_c
Hi,
I setup a SIP trunk between asterisk and Cisco
CallManager according the wiki page.
http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration
But I'm getting a 'Malformed/Missing URL' from the
CallManager. Does anyone know what went wrong here?
I'm running asterisk CVS HEAD and
What kind of card are they using? Is there only 1 telco circuit?
If so, then I'm thinking their card should have detected the loss of
service and switched to it's internal clock. Do they have a secondary
clock source available across another circuit? Perhaps a tie line to a pbx
that can be configu
Adam Robins asked:
Shouldn't [Asterisk] be smart enough to go to Zap/4 as the only available
port in the group [with a live trunk]?
Adam Goryachev wrote:
No, asterisk doesn't do dialtone detection.
But this isn't an issue of dialtone detection but one of detecting
battery (a much easier
Guys.,
How can I disable native briding on sip?
I get this but after that, the call just tries to do the bridge and freezes
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing Dial("SIP/demo-3763", "SIP/demo2|20|mwtWT") in new stack
-- Called demo2
-- Started music on hold
I think it would be interesting and not that difficult to adapt
Asterisk-Stat as an Employee Time Clock.
All it would require to calculate time difference between first and last
call on their extension with password if needed (and lunch brake if
needed).
When an employee comes to work he/she dials
I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk
with G729.
Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX
Hi guys
I am currently getting the following in my log
asterisk1*CLI> Parsing
'/etc/asterisk/manager.conf': Found
asterisk1*CLI> == Parsing
'/etc/asterisk/manager_custom.conf': Found
asterisk1*CLI> == Connect attempt from
'127.0.0.1' unable to authenticate
Can anyone tell
>
> >For some reason a couple weeks ago users began experiencing garbled
audio
> >in one direction when dialing out via our VoIP provider.
> >
> Play with the jitter buffer, I'll bet this is your problem. I had
> exactly the same problem with a cable ISP. Also, watch for strange
routing.
>
>
I can not get *77 to work on our Asterisk server .. @ home 1.1 final ...
Other * codes seem to work without issue .. Just can't use the *77 code ..
Anyone have any ideas what to look for ??
BRW
___
Asterisk-Users mailing list
Asterisk-Users@lists.digiu
Hi all,
One of the things that I keep being asked for is a sidecar for the
receptionist phone. Are there any SIP phones available on the market
with a sidecar in addition to the snom? Or is the snom my only option?
Any help would be appreciated.
Thanks,
Max
__
On Wed, 2005-06-22 at 17:49 -0400, Mike M wrote:
> On Wed, Jun 22, 2005 at 05:19:47PM -0300, Alessandro wrote:
> > But all ports are green!
>
> Really? Maybe they aren't making the RED FXO cards anymore. You should
> look at them carefully for p/n differences and not rely on colors. The
>
On Wed, 2005-06-22 at 11:46 -0400, Adam Robins wrote:
> I installed a TDM400P with 4 FXO modules. Before moving all of my
> office phone lines to it, I decided to move only one for testing. I
> plugged it into port 4 on the card.
>
> When I launch an outbound call as ZAP/g1/${EXTEN}, Asterisk go
This is not the fourm for AAH and AMP issues, please contant those fourms
On Thu, 23 Jun 2005, scott wrote:
Hi guys
I am currently getting the following in my log
asterisk1*CLI> Parsing '/etc/asterisk/manager.conf': Found
asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf':
Hi guys
I am currently getting the following in my log
asterisk1*CLI> Parsing '/etc/asterisk/manager.conf':
Found
asterisk1*CLI> == Parsing
'/etc/asterisk/manager_custom.conf': Found
asterisk1*CLI> == Connect attempt from
'127.0.0.1' unable to authenticate
Can anyone tell
Hi All,
Sorry for the double post, but I'm in a real bind.
I have several * servers connected to T1 PRI's from various service
providers in multiple locations the US. All the * servers use the same
hardware with the same OS and * version CVS-v1-0-11/09/04-12:27:27. When
connected to 5ESS Switch
Hello,
I'm curious if anyone has attempted using Asterisk with any Zaptel
cards in on one of IBM's OpenPower servers.
I've read via some googling and posts the the astmasters list that
there is/was a working version of the Zaptel driver for PPC under
Yellow Dog Linux.
Any thoughts?
Thanks,
Ilan
Rich Adamson wrote:
I have one POTS line going into a TDM400P. Here in Atlanta, we have 10 digit local
dialing. I launch a call "Zap/1/7705551212" and it goes thru just fine. The
next time I try it, without any modifications, I get a Bell recording telling me that I
must dial the area code
[EMAIL PROTECTED] wrote:
For some reason a couple weeks ago users began experiencing garbled audio
in one direction when dialing out via our VoIP provider.
Play with the jitter buffer, I'll bet this is your problem. I had
exactly the same problem with a cable ISP. Also, watch for strange rout
On 6/22/05, Memon, Nauman <[EMAIL PROTECTED]> wrote:
>
>
> I was told that the project has already been released in to the CVS head,
> and is available to us now, but not available yet for the business edition.
Nothing on the cvs mailing list as of yet.
-Brian
Hi all, just checkout the latest add-on and trying find how to use INSERT
cmd?
I got the MYSQL 5, is it possible to call a stored procedure from the mysql
add-on .
thx..
_
Powerful Parental Controls Let your child discover the bes
Shawn guessed correctly; "Most likely a channel bank with 24FXS." We have 2
cards each with 4 ports.
Ron
> [EMAIL PROTECTED] wrote:
>> We want to have the separate queues for tracking purposes but the queued
>> calls
>> need to be ordered and answered as if there was only one queue. For
>> exam
Hello.
I spoke with some of the support staff at Digium regarding
the Digium/Cepstral partnership. I was trying to find out when something may be
available, and if there is documentation about the project.
I was told that the project has already been released in to
the CVS head, an
Hello
I would like to know how can I manage to implement a table which translates
an extension number into a phone number. Let see an example:
If I dial an extension like 3021, Asterisk has to Dial an agent (our
employees) located at San Francisco using the following telephone number:
415
http://www.frsirt.com/english/advisories/2005/0851
A vulnerability was identified in Asterisk, which may be exploited by
authenticated attackers to execute arbitrary commands. This flaw is due
to a buffer overflow error in the manager interface that does not
properly handle specially crafted comma
On Wed, 2005-06-22 at 17:45 -0400, Joel Jn-Francois wrote:
> I have been trying to open an account with voipreach.net for over a week
> now and I have not gotten any response from them as yet. None of their
> phone numbers are working. Does anyone know if voipreach is still doing
> business?
Lee,
Please send e-mail to [EMAIL PROTECTED] and give me a call
604 780 2668?
Leon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Barken
Sent: June 22, 2005 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-
If I connect to a provider using iax, and that provider connects to
his provider using only sip, the provider I am connecting to isn't
going to be able to bridge the call and drop out of the media stream
correct?
If I'm understanding how bridging works, you lose the ability to have
the media stre
Dear List members,
I have an asterisk box whereon 45 GXP-2000 telephones from Grandstream are
connected at my work. This works fine.
Now I want to take 5 GXP-2000s to different homes on internet and want them
to be part of the same internal telephone system. One external GXP-2000 is
to be the nigh
On 6/23/05, Seamus Abshere <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] wrote:
> this is perhaps a silly question, but how do you have so many zaptel
> FXS's? do you have six TDM400 cards with four FXS's each? or what am I
> missing?
Most likely a channel bank with 24FXS.
__
All:
Any have sample config's for an ADIT600 w/ CMG card
connected to an Asterisk box via MGCP?
Thanks,
Garrett Smith
[EMAIL PROTECTED]
VoIPSupply.com
-- a division of B2 Technologies, LLC
(716) 250-3408 Direct
(716) 630-1548 Fax
(716) 903-9495 Cell
AOL IM:
Hi Alessandro,
> But all ports are green!
>
> p1 -green
>
> p2 - green
> p3 - green
> p4 - green
I think he means the daughter card color, not the LED on the card slot.
What color are the actual daughter cards?
J.
___
Asterisk-Users mailing lis
On Wed, Jun 22, 2005 at 05:19:47PM -0300, Alessandro wrote:
> Mike,
>
> I got current stable release in CVS repository, and I think that Ok.
> See below:
>
> /var/log/messages
> Jun 22 17:04:35 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0
> Jun 22 17:04:35 darthvaden kernel: PCI: S
I am trying to figure out a safe way to do 911 service. I would like
to have all my SIP phones register with a remote
Asterisk server and use it for inbound/outbound calls. However when
that server is no longer available or someone dials
911, i would like the SIP phones to fail over/use a
Hi,
Please see inline:
In Message-ID: <[EMAIL PROTECTED]>
Robert Goodyear <[EMAIL PROTECTED]> wrote :
>
> On Jun 22, 2005, at 1:50 PM, Zen Kato wrote:
>
> > Hi Robert,
> >
> >> Let me guess... mailbox 5103 or 5203 were the last in the list to
> >> receive it?
> >
> > Every trials(1-6) I got o
I have been trying to open an account with voipreach.net for over a week
now and I have not gotten any response from them as yet. None of their
phone numbers are working. Does anyone know if voipreach is still doing
business?
___
Asterisk-Users ma
I can do that. Please contact me off th email net. 626-814-2354 Michael
D. Schelin - ShellTel
Lee Barken wrote:
hi Leon,
We are initially looking for US only, but eventually would like to add
international toll free numbers. We would like inbound IAX2 or SIP.
Thanks,
-Lee
On Wed, 22
With a ISDN-32 (T1 in the US (and others ?), E1 in Europe and many other
places) you can have up to 32 channels.
I've got 4 E1's => Zap/120 works for me :)
Julian.
Seamus Abshere wrote:
[EMAIL PROTECTED] wrote:
We want to have the separate queues for tracking purposes but the
queued calls
n
New software released from my monkeys to your computer:
HooDaHek 0.1
Asterisk Caller ID Database, CGIs, and
Caller ID AOL Instant Messenger Bot
HooDaHek (hoo-dah-hek, as in 'who-the-heck?') is a collection of
Asterisk AGI scripts, CGI scripts, and MySQL tables intended to
implement your own in
hi Leon,
We are initially looking for US only, but eventually would like to add
international toll free numbers. We would like inbound IAX2 or SIP.
Thanks,
-Lee
On Wed, 22 Jun 2005, Leon Sun wrote:
> What kind of toll free do you need? For US only or whole North America?
>
> Do you need
On Wed, Jun 22, 2005 at 01:46:39PM -0700, Frank Mayhar said:
> On Wed, 2005-06-22 at 16:05 -0400, Walt Reed wrote:
> > (slow disk can cause high load average numbers as you spend
> > all your time in I/O Wait.)
>
> Um, no. At least in traditional Unix (meaning System V and the BSDs),
> the "load
On Wed, Jun 22, 2005 at 03:56:55PM -0500, Andrew Latham wrote:
> He is talking about ZAP channels. This is correct. No one should want
> to use that many ZAP channels.
You can use more than 250 zap channels. That limitation has been removed
a long time ago. chan_zap.c doesn't have to use the dev
> I have one POTS line going into a TDM400P. Here in Atlanta, we have 10
> digit local dialing. I launch a call "Zap/1/7705551212" and it goes
> thru just fine. The next time I try it, without any modifications, I
> get a Bell recording telling me that I must dial the area code and seven
> digit
On Jun 22, 2005, at 1:50 PM, Zen Kato wrote:
Hi Robert,
Let me guess... mailbox 5103 or 5203 were the last in the list to
receive it?
Every trials(1-6) I got only 51 mailboxes copied. My quick guess is
256/5(u0103 and &xx03s)=51...1, so changing tmp[256] to tmp[4096]
does not work. 'Pseudo-
On Wed, 2005-06-22 at 16:05 -0400, Walt Reed wrote:
> (slow disk can cause high load average numbers as you spend
> all your time in I/O Wait.)
Um, no. At least in traditional Unix (meaning System V and the BSDs),
the "load average" is the average length of the run queue. By
definition, if a pro
Hi Robert,
> Let me guess... mailbox 5103 or 5203 were the last in the list to
> receive it?
Every trials(1-6) I got only 51 mailboxes copied. My quick guess is
256/5(u0103 and &xx03s)=51...1, so changing tmp[256] to tmp[4096]
does not work. 'Pseudo-diagram' as you mentioned before(6/8/05)
is
Leonardo:
If you need a hand, only drop me an email.
Regards,
--
Ing CIP Alejandro Celi Mariátegui
<[EMAIL PROTECTED]>
El mié, 22-06-2005 a las 04:28, Leonardo F. Bauchwitz escribió:
> Hi:
> We have finished the translation of the FAQ of Digium to spanish.
> They are already (in Spanish) a
[EMAIL PROTECTED] wrote:
We want to have the separate queues for tracking purposes but the queued calls
need to be ordered and answered as if there was only one queue. For example,
if there are 3 calls in the helpdesk queue and 1 call in the isp queue, if a
new call comes in, no matter which que
Anyone out there tried going from cell to Motorola 02527MOT to FXO with
Asterisk?
What other kinds of fixed mobile service are available for use with
Asterisk?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailm
I have one POTS line going into a TDM400P. Here in Atlanta, we have 10
digit local dialing. I launch a call "Zap/1/7705551212" and it goes
thru just fine. The next time I try it, without any modifications, I
get a Bell recording telling me that I must dial the area code and seven
digit number
We have 1 queue called helpdesk and are setting up a second one called isp.
The helpdesk queue is for internal support calls and isp for our ISP customer
calls. Both of these queues will be directed to the same agents (helpdesk
phone extensions).
We want to have the separate queues for tracking
Mike,
I got current stable release in CVS repository, and I think that Ok. See below:
/var/log/messages
Jun 22 17:04:35 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0
Jun 22 17:04:35 darthvaden kernel: PCI: Sharing IRQ 9 with 00:1f.5
Jun 22 17:04:35 darthvaden kernel: Freshmaker
> Hello
>
> Maybe a silly question, but after some searching couldn't find answer. Is
> there a number I can dial to pickup and listen to my voicemail messages on
> my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages
> on Avaya system?
If you are using [EMAIL PROTECTED
Hi guys,
I have a weird thing happening sometimes with
users calling from a GrandStream phone through Asterisk onto a
PSTN.
Sometimes after a user hangs up a call on a
GrandStream phone the phone starts ringing after a couple seconds.
When the call is answered there is no one
there.
Anyo
> I need to try the CVS HEAD release of Zaptel on the stable Asterisk release.
> I was wondering how to do this. I currently have a stable Asterisk running
> on a Dell Server with Redhat Enterprise 3 Linux. I was wondering if anyone
> can give me some pointers of how to do this using CVS. Should I
I will need toll free for USA. If whle north america available, i would be
interested as well. the incoming call will come via SIP.
Thanks
Quoting Leon Sun <[EMAIL PROTECTED]>:
> What kind of toll free do you need? For US only or whole North America?
>
> Do you need carrier send incoming call
I doubt it's the software itself (I run Tiki too... It's just PHP.)
It's purely a matter of scaling. What part is causing the load? The PHP
apache processes? The DB server? Both? What performance tuning has been
done? Is it a custom apache compiled for this app or is it a generic
distro version wit
On Jun 22, 2005, at 2:07 AM, Zen Kato wrote:
Hi,
I also changed as following sequences;
app_voicemail.c
1. Line 3724 tmp[256] to tmp[4096] vm_exec
2. Line 3760 tmp[256] to tmp[4096] append_mailbox
3. Line 3796 tmp[256] to tmp[4096] vm_box_exists
4. Line 3290 tmp[256] to tmp[4096] vm_exec
Daniel,
we have the same problem when our PRI line drops and Zapras has to
reconnect. You will also notice that the pppd process does not die
when Zapras does and the ppp connection cannot re-establish itself.
What we normally do is restart asterisk and then kill the pppd process
with the comma
Make sure your fxo and fxs are in two different groups.
Otherwise, you won't be able to specify which one to steal.
Also, check out zapbarge, that should work better than meetme
for what you are trying to do.
Hope this helps,
Jon.
On Wednesday 22 June 2005 01:18 pm, Richard Koch wrote:
> Sor
Matt King wrote:
> The reason for this is that Orderly Software provides an advanced queue
> management system called OrderlyQ, that lets callers hang up and call back
> when they reach the front of the queue. OrderlyQ is patent-pending,
> and we do NOT allow the use of OrderlyCalls to provide si
On Wed, Jun 22, 2005 at 01:03:33PM -0300, Alessandro wrote:
> Mike,
> It's got 4 modules. What color are the modules in positions 1, 2, 3, 4
>
> on the TDM400P card? Don't be confused by the 0-3 numbering, just add
> 1.
>
> The colors in positions 3 and 4 are green, 1 and 2 light is off.
Y
Mike M <[EMAIL PROTECTED]> wrote:
> Think "opposite". Green modules are fxs and should be handled with the
> fxo signaling. Red modules are fxo and should be handled with fxs
> signaling.
>
> Note the red and green colors here:
> http://www.digium.com/index.php?menu=fxsvfxo
The opposite thing is
I need to try the CVS HEAD release of Zaptel on the stable Asterisk release.
I was wondering how to do this. I currently have a stable Asterisk running
on a Dell Server with Redhat Enterprise 3 Linux. I was wondering if anyone
can give me some pointers of how to do this using CVS. Should I rename t
I decided to test a similar scenario against another machine (server
C). This machine behaves in a similar way as server B. It is also
running on Gentoo. When I try to transfer a call into a conference
room, it fails. Below is the CLI output of an inbound call coming
from server A into serv
What kind of toll free do you need? For US only or whole North America?
Do you need carrier send incoming call to your Asterisk by SIP or by T1/E1
from Digium card?
Leon Sun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Barken
Sent: June 21, 200
Sorry about the lengthy post, I've searched high and lo for
information on how to do this but now I need your help...
Brief intro on problem and requirements ===
I'm hoping to use Asterisk in a Home environment where I'd like to
replace the current non-PC Answering Machine, an
As I understand it, the wiki software behind voip-info is not able to
keep up with the load. There may be better (-performing) alternatives
such as MediaWiki, but the question would be that of conversion... And
whether the owner even wants to convert.
> -Original Message-
> From: Bjørn Ov
modversions.h will be created when you recompile your kernel and
reboot with your new kernel, or install kernel-headers package with
the same version of your kernel.
Regards.
Elio Rojano
[EMAIL PROTECTED] wrote:
Hello,
I try to compile the driver zaptel and they give the following error:
li
Go to http://www.inaccessnetworks.com/projects/asterisk-oh323
Download oh323 0.65
Then go to
http://www.inaccessnetworks.com/asterisk-oh323/Libraries
download following
openh323-Janus_patch4-src-tar.gz (2555677 bytes)
pwlib-Janus_patch4-src-tar.gz (229 bytes)
Please read Readme from 0.65 car
That's what I was hoping someone else would say (:)
Thanks Tony.
Julian.
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>, Asterisk <[EMAIL PROTECTED]> wrote:
I've had several users today inform me that whilst they were on a call,
the volume kept fading in and out to such an extent
Correction 250 Zap channels.
Don this is not an issue for you.
On 6/22/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> look at ser projects:
> asterisk is limited to 250 channels
> You need cat 5e and manage qos if you setup ip phones
> Harry
> --- Don Brearley <[EMAIL PROTECTED]> a écrit :
>
>
Don Brearley wrote:
I do understand that I would need to replace all [300] of my existing
telephones with VoIP-capable
phones, and that I'll need to re-wire most of the campus telephone
infrastructure (it's still
all cat-3) -- these arent problems.
dear Don & the rest,
Imagine:
*replacing
Hi!
I've a problem with caller presentation, when I call a number my
presentation is set to "Presentation permitted, user number passed
network screening", this works for outgoing call but not for incoming,
numbers are always hidden and network provided number is always screened.
I don't unde
> > I do understand that I would need to replace all of
> > my existing telephones with VoIP-capable
> > phones, and that I'll need to re-wire most of the
> > campus telephone infrastructure (it's still
> > all cat-3) -- these arent problems.
why do you think that you need to do that ?
you could
I guess that my definition of "first available trunk" (either forward or
backward) differs from Digium. I would think that the card should know
which ports had an electical signal attached.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Kawakami
Sen
9- Loaded the xcfxs driver
modprobe wcfxs
I think that you are using wrong driver!! Check for something like
'wctdm'!!!
Best Regards,
Anatoliy Kounitskiy
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On Wed, Jun 22, 2005 at 05:32:22PM +0200, harry gaillac wrote:
> look at ser projects:
> asterisk is limited to 250 channels
What kind of crack are you smoking? There are people that have set
up more than 250 channel systems.
Matthew Fredrickson
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A
> I've been planning to replace my aging CENTREX switch with a new PBX and am
> seriously
> considering Asterisk as my solution.
Great
> I work at a college, and we currently support just under 300 regular analog
> lines to
> the offices and whatnot.
Free help then
> I was wondering.. Is
You need your kernel-headers.
Sebas
[EMAIL PROTECTED] wrote:
Hello,
I try to compile the driver zaptel and they give the following error:
linux01:/usr/src/zaptel# make install
gcc -Iir/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Iir/drivers/
l -I. -Wstrict-prototypes -fomit-frame-pointe
-Original Message-
Message: 18
When I launch an outbound call as ZAP/g1/${EXTEN}, Asterisk goes to
Zap/1 and I hear dead air because there is no line attached to that
port. Shouldn't it be smart enough to go to Zap/4 as the only available
port in the group?
-you obviously read the wi
Hello
Maybe a silly question, but after some searching
couldn't find answer. Is there a number I can dial to pickup and listen to
my voicemail messages on my SIP phone? I am used to eg dialling *17 to
pickup my voicemail messages on Avaya system?
Angus
In article <[EMAIL PROTECTED]>, Asterisk <[EMAIL PROTECTED]> wrote:
> I've had several users today inform me that whilst they were on a call,
> the volume kept fading in and out to such an extent that they thought
> the caller had hung up.
>
> I would dismiss this if it were a single person ment
Hi all,
A new version of DIAX is available for download: 0.9.15a.
For the moment you can find it only at the following location:
http://www.cosmica.ro/dante
and
http://www.geocities.com/tdanro
Whats new in this version comparing with 0.9.10f (the latest official
version):
- GSM/PSTN Gateway fu
http://voip-info.org/tiki-index.php?page=Asterisk+manager+api
On 6/22/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi
> One question for you!
> Which operation are allows by Asterisk Manager API?
> Can I connect to Asterisk server, create a new Channel,add channel on Asterisk
> server,specif
On 6/22/05, Peter Svensson <[EMAIL PROTECTED]> wrote:
> Grandstream clains thay will address the speakerphone problems in an
> upcoming release. I think they need a more advanced echo canceler since
> the speaker and microphone are acoustically strongly coupled.
This is the same problem that all t
So, instead of the agent dialling in and logging on, and waiting for an
inbound call (therefore cannot make outbound calls, and possibly
incurring call costs from their line to the PBX), could you not get them
to login via agentcallbacklogin, which then drops the line. If they want
to make an
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