John,
For incoming fax numbers, did you port existing numbers or did you get new
numbers from bandwidth.com ?
If the later, what if you switch for another provider ?
Would you then be able to port the given number to your new provider ?
Regards
___
how to loging agent asterisk 1.4.13?
thanks
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On 10/10/07, Ex Vito [EMAIL PROTECTED] wrote:
Hi list,
I'm evaluating a private telephony scenario of about 20
locations - 300 phones, 50 FAX machines.
More than 1 PRI?
All other locations, small by themselves, would get SIP
phones managed by asterisk, since there is good IP
Hello Ken,
Hi, all. My company is setting up a branch office in Germany, and I'm
very interested in a VoIP provider over thataway.
as I am living in Germany, let me advise you against using VoIP
providers in Germany. Most of the time they do work, but they are not
as reliable as a regular
Hijacking a thread again - the only way I can post to the -user list is
by replying to another thread. (btw, this is getting really annoying.
Please, please, please, Digium, sort the filters out!)
I've added the auto-answer header in my dialplan, and it works great.
However, there is a problem
VOIP Users Conference Friday Oct 12 @ 12:30 PM EDT
Today, besides whatever else is on anyone else's minds, we'll be
discussing some asterisk communication techniques for use with SMS,
email and web API.
Also, some typical uses for the asteriskdb, what it is and why you
care about it at all.
Hello,
I wonder if there is a way to build my own asterisk application (let us say
apps/app_myappl.c),
and to launch other existing applications from it (for example, doing an
apps/app_dial.c, or others).
Could someone highlight me on that?
thx
Pirlouwi.
On Friday 12 October 2007 04:28:42 Pirlouwi wrote:
I wonder if there is a way to build my own asterisk application (let us say
apps/app_myappl.c),
and to launch other existing applications from it (for example, doing an
apps/app_dial.c, or others).
Both the Page app and VoicemailMain do this,
On 10/12/07, Yair Hakak [EMAIL PROTECTED] wrote:
you'll have to excuse the ignorance (i'm a software guy, not a telcom
guy..)
Is there any way to know if a channel has been answered by an automatic
system (like voicemail) rather than a human being?
Specifically, I want to use a .call to
No, I am not sure whether it's still an issue with the newer series because
I have not had a chance to test this with one of those models.
Regards,
Joseph
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield
Sent: Thursday, October 11, 2007 4:35 PM
To: Asterisk
Is there an example on how to use two E1 ports connected to each other
to simulate connections? Since I do not have an E1 at the office I need
for one port to act normally and the other to act as if it were the
telephone company so I can send calls from one E1 to the other. Someone
has
Hi Steve,
you are totally right, but my question is because a saw that gui into SVN
and not yet released, but at the same time used into AsteriskNOW.
Was just a question...
;o)
Thanks
2007/10/12, Steve Totaro [EMAIL PROTECTED]:
FaberK wrote:
Hi to all,
I've just started to see that
In 'top', you can always look at what percentage of your CPU is idle.
Subtract that from 100 and you've got your load average.
Cpu(s): 1.1% us, 0.6% sy, 0.0% ni, *98.1% id*, 0.1% wa, 0.1% hi,
0.0% si
Erik Anderson wrote:
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I don't
Hi,
I am going to setting up an asterisk setup similar to yours. I talked to the
folks at bandwidth.com and they SIP trunking for $25/mo/line with 2000
minutes per line. I haven't tried their service yet though, so I can't say
how the quality is.
Regards,
Zaheer
-Original Message-
From:
Crossover cable and pri_net/pri_cpe. It is well documented if you
search a little.
Thanks,
Steve Totaro
Carlos Chavez wrote:
Is there an example on how to use two E1 ports connected to each other
to simulate connections? Since I do not have an E1 at the office I need
for one port to
Vincent wrote:
4. install ztdummy:
echo ztdummy /etc/modules
modprobe ztdummy
===
Since you are using the OpenVOX FXO card, don't you need another
module? I'm guessing you'd need wctdm INSTEAD of ztdummy.
___
--Bandwidth and
If you read between my lines, my guess is it won't be released unless
the community takes it out of Digium's hands and forks it.
Thanks,
Steve
FaberK wrote:
Hi Steve,
you are totally right, but my question is because a saw that gui into
SVN and not yet released, but at the same time used
Steve Totaro wrote:
I don't think that is correct. I am running worldcommunitygrid and this
is what I get
top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02
Cpu0:0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st
On Friday 12 October 2007 10:45:52 Steve Totaro wrote:
FaberK wrote:
I've just started to see that Asterisk-gui from Digium.
Does anybody know, when the first official-realese will be released?
I may be totally wrong but at Astricon, during the What's New at
Digium (SwitchVox purchase) I
Does anyone have any tricks to allow codecs based on what network a phone is
on? i.e., allow uLaw if the device is on the LAN, and only allow g729 if
the device is anywhere else?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836
I don't think that is correct. I am running worldcommunitygrid and this
is what I get
top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02
Cpu0:0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st
Cpu1:0.0%us,0.0%sy,100.0%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st
On 10/12/07, D4rk F1ber [EMAIL PROTECTED] wrote:
Curious what others are using, and if anyone can make some
recommendations? Not sure if this has been covered already on the
list, and not sure if recommending companies are allowed, so maybe I
need get replies off list?
There are quite
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I don't think there is a formula like
cpu usage = loadavg / #cpus
A loadavg of 3 says that there are 3 processes waiting to
be executed.
Anyway, I'll admit that a loadavg of 3 /might/ be ok.
Here's a quote from this page:
Yes that is correct. I have no analog phone service here, I plan on using
direct SIP Trunking. My connections to the internet are two load balanced
high-speed connections. The first is a cable modem operating at 20d/5u and
the second is a fiber optic connection that is operating at 20d/5u. Both
Erik Anderson wrote:
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I wouldn't be too happy about a system with a
loadavg of 3.
The system he mentioned had 8 cores, though. So a load average of 3
is less than 50% usage.
I don't think there is a formula like
cpu usage = loadavg /
Hi List;
When I readed about DUNDI, I though there is a website
and we can buy and sell routes there using DUNDI, but
when I tried to browse http://www.dundi.com) then I
found a strange page that contains only this text It
Works!
So, what is the market of DUNDI really? Is there a
method to buy
Is there an easy way to show all active channels AND the codecs
they're using? Other than going through EACH channel individually to
check the codec which is, obviously, not a very efficient process.
Thanks,
Scott
___
--Bandwidth and Colocation
Zaheer Master wrote:
AsteriskNow running one of two ways:
1) As a virtual machine on a VMWare server (Eight core Xeon server
with 4GB
ram)
You wouldn't be able to add TDM cards to this scenario if you wanted to
down the line, as vmware can't access the hardware installed in a box.
On Fri, 12 Oct 2007, Tilghman Lesher wrote:
On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote:
Atis Lezdins wrote:
I have 8-core system that has web interface + sql + java + some other
stuff running, and at 30 simultenous calls i get loadavg maximum of 3.
I wouldn't be too happy
Eric ManxPower Wieling wrote:
Julian Lyndon-Smith wrote:
Hijacking a thread again - the only way I can post to the -user list is
by replying to another thread. (btw, this is getting really annoying.
Please, please, please, Digium, sort the filters out!)
You seem to be subscribed to the
Olivier wrote:
2007/10/12, Jonn Taylor [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Would you then be able to port the given number to your new
provider ?
Yes, if you are in the US. Number port ability is one thing that ALL
VIOP providers had to provide.
Here too
I was told yesterday (by Cantata guy) that T.38 demands a good level of QoS.
That surprised me a lot as I thought the whole purpose of T.38 was to avoid
SIP and ToIP latency.
Another editor (Interstar) told me T.38 passthrough doesn't work.
As devil lies in details and I couldn't get any, I'm
Hi, Brian:
Brian West wrote:
Matt,
I talk very openly about this issue. It was very rude of you to bring
this up. This plea was total bullshit. If you want to know the whole
story feel free to call me and talk about it. 918-424-9378... anyone
can call me and ask me questions about it.
2007/10/12, Jonn Taylor [EMAIL PROTECTED]:
Would you then be able to port the given number to your new provider ?
Yes, if you are in the US. Number port ability is one thing that ALL
VIOP providers had to provide.
Here too (France), number portability is mandatory but in facts, I couldn't
Atis Lezdins wrote:
I have 8-core system that has web interface + sql + java + some other stuff
running, and at 30 simultenous calls i get loadavg maximum of 3.
I wouldn't be too happy about a system with a
loadavg of 3.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566
Search for Agent at http://www.voip-info.org/
On 10/12/07, Walter Willis [EMAIL PROTECTED] wrote:
how to loging agent asterisk 1.4.13?
thanks
___
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Hi Zaheer,
Go the standalone machine. It will save you time and effort.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi to all,
I've just started to see that Asterisk-gui from Digium.
Does anybody know, when the first official-realese will be released?
Thanks to all
--
.:FaberK:.
___
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asterisk-users
On Friday 12 October 2007 04:38, Ken D'Ambrosio wrote:
Hi, all. My company is setting up a branch office in Germany, and I'm
very interested in a VoIP provider over thataway. However, I'd need a few
things:
- Reliability. Can't have my branch office's DID's just going down. A
then you
I've been trying to post this message to the list, but it keeps getting
bounced back to me. Sorry if duplicate messages are coming through.
Hi All,
I have done some research on Asterisk and I would like to try it in my
office. Here's what I'm looking at for my system:
AsteriskNow
Yes, this is true when using presence on the 601's. With presence disabled,
you get no reboots at all. That's why when I realized that, I decided on
the setup I mentioned below. However, you could let us know whether this is
true for the 650's 550's and 330's.
Joseph
From: [EMAIL
Pirlouwi wrote:
Hello,
I wonder if there is a way to build my own asterisk application (let us
say apps/app_myappl.c),
and to launch other existing applications from it (for example, doing an
apps/app_dial.c, or others).
Could someone highlight me on that?
thx
Pirlouwi.
Even better
Hi All,
I have done some research on Asterisk and I would like to try it in my
office. Here's what I'm looking at for my system:
AsteriskNow running one of two ways:
1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB
ram)
2) on a P4 2.4Ghz with 768mb RAM
I'm
Newbie to the list. Are the archives searchable like Listserve
archives, or do you just use Google? In another reality I am a
mainframe Sysprog, so this stuff is somewhat alien to me.
I would like to find previous postings on running Asterisk with CoLinux.
I have the Win32 version of
I use a mysql script to dynamically generate the page command and
page about 70 phones, and I have never had a reboot problem.
Sometimes there is a slight delay waiting for all the phones to join
the page conference. I am using a mix of 650's, 550's, and 330's.
It must only be an issue
On Friday 12 October 2007 11:10:02 Gordon Henderson wrote:
On Fri, 12 Oct 2007, Tilghman Lesher wrote:
On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote:
Atis Lezdins wrote:
I have 8-core system that has web interface + sql + java + some other
stuff running, and at 30 simultenous
Stephen Bosch wrote:
Philipp Kempgen wrote:
Don Kelly wrote:
http://www.sandman.com/autodial.html
These phones look like the ones we had in Germany
20 years ago. ;-P
Hey, don't knock it, Phillipp :) -- I'm as big a fan of German
technology as anybody, but these phones are amazing
Olivier wrote:
John,
For incoming fax numbers, did you port existing numbers or did you get
new numbers from bandwidth.com http://bandwidth.com ?
Both, we ported numbers and got new one's.
If the later, what if you switch for another provider ?
I did alot of research before we went with
hi all,
you'll have to excuse the ignorance (i'm a software guy, not a telcom
guy..)
Is there any way to know if a channel has been answered by an automatic
system (like voicemail) rather than a human being?
Specifically, I want to use a .call to make a call on a channel and only do
something
On Fri, Oct 12, 2007 at 11:28:42AM +0200, Pirlouwi wrote:
Hello,
I wonder if there is a way to build my own asterisk application (let us say
apps/app_myappl.c),
and to launch other existing applications from it (for example, doing an
apps/app_dial.c, or others).
Could someone highlight me
Hi Everyone,
Sorry in advance if this not the correct place to ask this question,
feel free to point me somewhere more appropriate to ask.
We have an Asterisk 1.2.7.1 server (about a year old version of
Asterisk @ Home with FreePBX) running the phone system for our small
office (roughly 15
On 10/12/07, D4rk F1ber wrote:
So I have my asterisk box up and working internally at home and all is
good so far. The next thing I wanted to do was make and recieve calls
to regular land lines now.
I don't have a POTS line and was looking for probably a SIP trunk. [...]
There are
Peer Oliver Schmidt schrieb:
...
as I am living in Germany, let me advise you against using VoIP
providers in Germany. Most of the time they do work, but they are not
as reliable as a regular phone company.
Hi,
on the one hand, I should ignore this thread, because it is not
asterisk
Hi,
am building the latest version of Asterisk (1.4.13) on a self-build
Linux host (based on LFS-6.3).
Version 1.4.12 built installed and worked fine. Last night I upgraded
the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1
against it. That seemed to build and install O.K.
What a waste of time...
dave
On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch wrote:
Brian West wrote:
And what was the purpose of this?
So that we would realize who we were talking to.
:)
-Stephen-
___
--Bandwidth and Colocation
On Friday 12 October 2007 10:34:39 C. Duncan Hudson wrote:
I'm fairly new to Asterisk, so please bear with me if this is silly
question. I'd like to run a script on my server that would take the
Call now to order banner off my website automatically when I put my
phone system on night. I can
On Friday 12 October 2007 10:29:24 Philipp Kempgen wrote:
Atis Lezdins wrote:
I have 8-core system that has web interface + sql + java + some other
stuff running, and at 30 simultenous calls i get loadavg maximum of 3.
I wouldn't be too happy about a system with a
loadavg of 3.
I dunno, 3
Brian West wrote:
And what was the purpose of this?
So that we would realize who we were talking to.
:)
-Stephen-
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To UNSUBSCRIBE or update options
Using two Asterisk connected between they, How do I can check the voicemail in
a remote system but working like *97?
I mean dont want ask the voicemail box, just the password and go to the
voicemail of caller. If I have the same extensions in the two Asterisk it
doesn't work.
Thanks.
--
Thanks for the quick response Dean. Can you elaborate any more? The reason I
would like to use the virtual machine is that I can dynamically allocate
more resources to the asterisk server as needed.
Regards,
Zaheer
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I guess it all depends on how you put your system on night mode. If you use
asterisk's database, and manually put it in night mode every day, my guess
is that you dial an extension which puts it in nightmode. You could include
as part of this the system command to touch a file, and then on your
Well, this has become a hot topic! :p
Thinking about my original post, I was reluctant of installing my PBX on a
shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual Core CPUs
@2GHz (4 totals cores) and 4GB RAM which serves as Domain Controller and
File Server (Samba), central backup
Hi All,
I have a quick one for you. Is there a way to mask
(i.e. combine) the flags in the Dial() application? In
other words, a way to do something like
Dial(Zap/1,10,d|t|f)
to get the effects of the three flags together in one
shot? I have a need to combine the effects of the o
and A flags in
Hello
1. I don't have deep knowledge of either Linux or Asterisk, but I seem
to have successfully installed 1.4 with Zaptel (for support for an
OpenVox PCI FXO card) on a stock Ubuntu 7.04 Server Edition:
dmesg ==
[ 25.990943] Zapata Telephony Interface Registered on major 196
Sorry...I should have been more specific in my original reply.
In 'top', you can always look at what percentage of your CPU is
idle. Subtract that from 100 and you've got your load average.
I should have said you get your average load percentage, rather than
just average load.
Mik Cheez
Is the call being dropped or is Asterisk takng a core dump?
I have core dump issues with g729 and asterisk 1.4.11, but my set up is
a little different than yours...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Moseman
Sent: Friday,
Hello My Aster-Friends!
I would like to hear if anyone out there in
Asteriskland has used the Dock-N-Talk (DNT) box to
connect cell phones to Asterisk box.
I have a couple of these boxes that I need to make
work with Asterisk, connected with Digium TDM400P
card. Anyone tried it before, and how
I use a mysql script to dynamically generate the page command
and page about 70 phones, and I have never had a reboot problem.
Sometimes there is a slight delay waiting for all the phones to
join the page conference. I am using a mix of 650's, 550's, and
330's.
It must only be an issue if
Search for Queue at http://www.voip-info.org/
On 10/12/07, Walter Willis [EMAIL PROTECTED] wrote:
HOw to call queues in asterisk ?
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To UNSUBSCRIBE
Julian Lyndon-Smith wrote:
Hijacking a thread again - the only way I can post to the -user list is
by replying to another thread. (btw, this is getting really annoying.
Please, please, please, Digium, sort the filters out!)
You seem to be subscribed to the list as [EMAIL PROTECTED]. Is that
I'm fairly new to Asterisk, so please bear with me if this is silly
question. I'd like to run a script on my server that would take the
Call now to order banner off my website automatically when I put my
phone system on night. I can handle the webserver side of things, but I
don't know where
FaberK wrote:
Hi to all,
I've just started to see that Asterisk-gui from Digium.
Does anybody know, when the first official-realese will be released?
Thanks to all
--
.:FaberK:.
I may be totally wrong but at Astricon, during the What's New at
Digium (SwitchVox purchase) I asked the
C. Duncan Hudson wrote:
I'm fairly new to Asterisk, so please bear with me if this is silly
question. I'd like to run a script on my server that would take the
Call now to order banner off my website automatically when I put my
phone system on night. I can handle the webserver side of
No ideas on this one from anyone? I suppose I'm going to need to pay
for some Digium support because this is a really unusual problem.
Does anyone else have a gateway that speaks g729 to Asterisk and
works? For whatever reason, Asterisk refuses to reply back to any of
my gateways using g729.
Not at all! You apparently don't realize (sic) you're talking to is just
a subtle way of saying look me up! Matt was just nice enough to do the
leg work :-)
On 10/12/07, David Boyd [EMAIL PROTECTED] wrote:
What a waste of time...
dave
On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch
On Friday 12 October 2007 01:33:44 pm Baji Panchumarti wrote:
On 10/12/07, D4rk F1ber wrote:
So I have my asterisk box up and working internally at home and all is
good so far. The next thing I wanted to do was make and recieve calls
to regular land lines now.
I don't have a POTS
So I have my asterisk box up and working internally at home and all is
good so far. The next thing I wanted to do was make and recieve calls
to regular land lines now.
I don't have a POTS line and was looking for probably a SIP trunk.
I have seen mentions of Skype integration with Asterisk, but
Actually, that looks right...look at your load average...
Steve Totaro wrote:
I don't think that is correct. I am running worldcommunitygrid and this
is what I get
top - 13:18:56 up 3 days, 22:49, 1 user, load average: 4.00, 4.04, 4.02
You mean like:
Dial(Zap/1,10,dtf)
?
On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote:
Hi All,
I have a quick one for you. Is there a way to mask
(i.e. combine) the flags in the Dial() application? In
other words, a way to do something like
Dial(Zap/1,10,d|t|f)
to get the effects of the
Is it possible that you are adding this, and then freePBX is overwriting
your file, in effect, taking out your addition?
IIRC, the newer versions of freePBX have the ability to hide users. I
wouldn't bet any money on that recollection, though.
On 10/12/07, Jesse Scott [EMAIL PROTECTED] wrote:
Worked fine for me.
On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote:
Hello My Aster-Friends!
I would like to hear if anyone out there in
Asteriskland has used the Dock-N-Talk (DNT) box to
connect cell phones to Asterisk box.
I have a couple of these boxes that I need to make
work with
How do you get 11ms translation time on ulaw 729 ?
we have 12ms and its dual xeons 2.6..
On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote:
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I wouldn't be too happy about a system with a
loadavg of 3.
The system he mentioned had 8 cores, though. So a load average of 3
is less than 50% usage.
-erik
___
--Bandwidth and Colocation
Gateway sends Asterisk an INVITE (using g729)
Asterisk sends Phone an INVITE (using g711 or g729)
Phone sends Asterisk an OK (using g711)
Asterisk sends Gateway an OK (with no RTP choice)
Gateways ends the conversation
I can setup the Phone to use g729 and it will reply with an OK for
g729, but
If you are going the external app method why not fire a script that
updates a DB?
Lacy Moore wrote:
I guess it all depends on how you put your system on night mode. If
you use asterisk's database, and manually put it in night mode every
day, my guess is that you dial an extension which
The cards ship configured for T1. If you didn?t change the jumpers, it is
set for T1.
If it is set for T1 and you really want an E1 and you configure your
zapata.conf as you would for an E1, you will get an error around channel 25,
which tells you that you forgot to change the jumpers, and you
That was actually a VM. Here's the real server (13ms).
CLI show translation
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
ulaw- 3-12 21 3 13 - 152-
alaw- 31-2 21 3
How many licenses you have (show g729 should give you this info)
Scott Moseman wrote:
Gateway sends Asterisk an INVITE (using g729)
Asterisk sends Phone an INVITE (using g711 or g729)
Phone sends Asterisk an OK (using g711)
Asterisk sends Gateway an OK (with no RTP choice)
Gateways ends the
Hi,
Check if you have a ground loop.
If yes, this is probably the cause of this hum.
Open the loop.
Best Regards,
Francois BERGERET
France
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To
Doesn't look like FreePBX is nuking it. I just SSH'd in and opened
the voicemail.conf directly and the entry is still in there:
305 = 1234,Users Name,[EMAIL PROTECTED],,attach=yes|saycid=yes|
envelope=yes|delete=no|hidefromdir=yes
I tried issuing both a 'reload' and a 'restart gracefully'
If you are using freepbx, I think freepbx actually simulates the
app_directory, so you may have to do something in the gui to fix it,
or they may not have such an option.
Hope this helps.
on Friday 10/12/2007 Jesse Scott([EMAIL PROTECTED]) wrote
Hi Everyone,
Sorry in advance if this not
Maybe I'm wrong, but don't you have to stop/start asterisk for
voicemail changes to take effect on 1.2 (like zapata)
Matt
On 12/10/2007, Jesse Scott [EMAIL PROTECTED] wrote:
Doesn't look like FreePBX is nuking it. I just SSH'd in and opened the
voicemail.conf directly and the entry is still
I agree with the suggestions of Teliax, and I also use
http://vitelity.net/ -- you can rent 800# DIDs from them for
$0.50/month plus minutes, and I think their US minutes are like $0.02 or so
Moj
D4rk F1ber wrote:
So I have my asterisk box up and working internally at home and all is
good
Hi,
I am looking for a syntax highlighter for AEL2. Google is not helping,
so I thought you guys could help me.
I found this vim syntax highlighter for AEL but it doesn't help if you
want to code in AEL2:
http://vim.sourceforge.net/scripts/script.php?script_id=1900
Cheers,
PLL.
How about the Intellitouch XLink? www.xlinkgateway.com
http://www.xlinkgateway.com/
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore
Sent: Friday,
On Friday 12 October 2007 17:14:39 Matt Gibson wrote:
Maybe I'm wrong, but don't you have to stop/start asterisk for
voicemail changes to take effect on 1.2 (like zapata)
Nope, reload.
--
Tilghman
___
--Bandwidth and Colocation Provided by
On Friday 12 October 2007 15:47:41 Anthony Francis wrote:
Lacy Moore wrote:
I guess it all depends on how you put your system on night mode. If
you use asterisk's database, and manually put it in night mode every
day, my guess is that you dial an extension which puts it in
nightmode.
How about the Intellitouch XLink? www.xlinkgateway.com
http://www.xlinkgateway.com/
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore
Sent: Friday,
Ok, I commented out the entry from voicemail.conf and that person is
gone from the directory.
However, uncommented with hidefromdir=yes in the options, they show up.
I just upgraded FreePBX to 2.3.0 and I couldn't find any additional
options for hiding the user from the directory.
This
I've used Vitelity.net for several years and am reasonably happy with
them.
You can get DID's from any area code you want for $1.49 a month with per
minute rates between $0.011 and $0.0149.
I also use Nufone.net for outbound. Their DID's cost $5.00 a month so I
don't have one :)
Nufone let's
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