Smartass... :)
Trixbox works off FreePBX which, while not as tightly integrated into
Asterisk, is currently far more mature and easy to use.
Note the use of the word currently. :) I wouldn't be too surprised
if FreePBX made the move to the Asteri/s/kNow framework. Removes the
big ugly
There is currently a reimplementation of zaphfc (vzaphfc). Perhaps it is
more robust.
Which package include such reimplementation of zaphfc ?
Thanks
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Hello,
I see that you are using T option (allow the /calling/ user to
transfer the call) when dialling to internal extensions and t (allow
the /called/ user to transfer the call) when receiving calls (in home
context). This it is why inbound transfer works fine and only one time.
So, I
Hi,
I configured Asterisk to run as asterisk user, but I see that a user can
anyway get a root sheet using !command in CLI. I understood that it's
something related to safe_asterisk and TTY console, but modifying the script
safe_asterisk I wasn't able to disable this root access.
Can someone
What i need is to recieve a call in a console!
I mean i can call from CLI...but can i recieve calls too? If this is possible
how is the console identificated and where!
Actually i need to call from one Asterisc server console to another(i know
what is asterisc server for, but this is a
I had the same issue when the us enterd a response. I used the read cmd, set
a variable and then used some gotoif's
- Original Message -
From: Yuan LIU [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 12, 2007 6:49 AM
Subject: [asterisk-users] Extensions in
This is real simple. in sip.conf do: callerid=Bjørn, Marius 966
do this setting for all the sip accounts that belong to 966
- Original Message -
From: Bjørn Marius [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
we've just released Asterisk-Java 0.3-m2 at http://asterisk-java.org.
The Asterisk-Java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk PBX
Server. Asterisk-Java supports
Le 11 févr. 07 à 21:25, Tzafrir Cohen a écrit :
On Sun, Feb 11, 2007 at 08:57:30PM +0100, Olivier MONNET wrote:
Hi,
I’m using Bristuff for more than a year and half now, and I am stuck
with the same problem since asterisk 1.2.
When using a card from Junghans, QuadBri or OctoBri everything is
On Mon, Feb 12, 2007 at 10:36:51AM +0100, jeremij jerome wrote:
Hi,
I configured Asterisk to run as asterisk user, but I see that a user can
anyway get a root sheet using !command in CLI. I understood that it's
something related to safe_asterisk and TTY console, but modifying the script
On Mon, Feb 12, 2007 at 09:15:01AM +0100, Olivier wrote:
There is currently a reimplementation of zaphfc (vzaphfc). Perhaps it is
more robust.
Which package include such reimplementation of zaphfc ?
Thanks
Currently the only public repository for it is the Debian package :-(
svn co
Try changing the shell for the asterisk user to /bin/false. This should
disallow anything passed through the ! command since it runs the command
via the shell for the asterisk user.
jeremij jerome wrote:
Hi,
I configured Asterisk to run as asterisk user, but I see that a user
can anyway
So here's my questions then. If APIC routes the IRQs to 1-15 for real
world usecan you safely have two devices on, say, 14? APIC will
assign one to maybe 23 and one to 20. But are they really both on 15
with a potential for conflict?
The conflict only happens if your OS is not APIC
Imagine that situation:
User 100 is speaking with user 150.
I am the 105 user. In my phone (105) I press *98100 (for example) and
now the user 100 is empty and user 150 is speaking with me.
Someone knows what I must to do???
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On 2/12/07, Paradise Dove [EMAIL PROTECTED] wrote:
my card has just fxo modules and is put in a 3.3v slot.
when running modprobe wctdm24xxp
it waits for ever and dmesg shows Resetting the modules
what could be the problem?
when i put this card in another system with 5v slot it works fine.
Thanks all for your input.
Based on the comments given I guess I could
replace the X100p for a TDM22B and then connect the Fax machine to the
TDM22B
and in the dial plan hae
exten = fax,1,Dial(SIP/40) (being the TDM port) and then it will ring
the fax machine
and pass the call in from the PSTN
Hi
i have a big problems with my asterisk .. i use a Digium TDM400P for
connect a
analog line.
And not all time (i don't know why) he don't see the end of the call and
anyone can call me
(occuped)
For that's work, i am disconnect the phone cable and it's good
anyone have a idea ?
bye
Hi guys,
I have the following configuration:
10 SIP softphones -- Asterisk -- PSTN
Audio is always good on SIP softphone side, but callers from PSTN side
*sometimes* complain that the audio quality is bad (and volume low). The
QoS is turned on on the computers where SIP softphone is installed,
I have been handed three Quintum tenor AX gateways which I am suppose
to configure for use with our soon to be deployed Asterisk 1.4 system.
Through some mix-up we only have hardware support even though the boxes
are brand new. We are working on getting software support.
I would like to
If the PSTN side is only complaining about conversations with a single phone
on the SIP side, look at the SIP phone.
Check the settings for that SIP phone/PC (VAD disabled, NIC settings,
runaway processes). Do PSTN callers here choppiness from the SIP phone
caller?
-Original Message-
Is it still needed to run fxotune on the TDM24XXE cards with
hardware echo cancellation?
Jerry
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Well, the PSTN side is complaining about a random phone on the SIP side.
Yes, they do hear choppiness.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Monday, February 12, 2007 2:49 PM
To: 'Asterisk Users Mailing List -
If it's a random phone on the SIP side, we have to look further upstream.
While jitterbuffers may help, in my opinion they mask a problem.
What type of connection do you have to the internet? Have you done
tracert's to your voip provider? What do they look like?
When you say that you do QoS -
Really? It's 9:23am EST and they aren't open yet.
I would call Digium's tech support. They open in 20 minutes.
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Do you know if it is possible to handle some events with phpagi?
For example:
On hangup (doesn't care if by caller or by asterisk) do something
Thanks
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On 2/12/07, Mohamed Farid [EMAIL PROTECTED] wrote:
I have a Cisco Call Manager - and need to use the IVR Feature from
Asterisk.
My extension is 400 and I am calling 558 on Asterisk
In my extension.conf I have these lines :
exten = 558,1,Answer
exten = 558,2,Playback(message.wav)
exten =
On Mon, 12 Feb 2007, Barry Fawthrop wrote:
Thanks all for your input.
Based on the comments given I guess I could
replace the X100p for a TDM22B and then connect the Fax machine to the TDM22B
and in the dial plan hae
exten = fax,1,Dial(SIP/40) (being the TDM port) and then it will ring the
fax
From what I read on the voip-info page, they did not define how to have
the phone muted when a paging call would go out. I already have the
paging (without mute) working using those same headers from that site.
But they don't cover the issue I'm trying to solve with my Polycom
Soundpoint 501s and
Usually you should use the manager interface for that.
On 2/12/07, nik600 [EMAIL PROTECTED] wrote:
Do you know if it is possible to handle some events with phpagi?
For example:
On hangup (doesn't care if by caller or by asterisk) do something
Thanks
I'm going to need to build a few Asterisk boxes that have dual and quad T1
interfaces. I knew Digium made T1 interface cards and on this list I heard
about Sangoma so I did a quick search and found the hardware page at
voip-info.org which lists several manufactures I didn't know about. All that
You have people administering your asterisk server who you wouldn't trust
with access to the machine? EEEK.
On 2/12/07, jeremij jerome [EMAIL PROTECTED] wrote:
Hi,
I configured Asterisk to run as asterisk user, but I see that a user can
anyway get a root sheet using !command in CLI. I
I recommend you to use Sangoma A102D or A104D.
Regards,
Radu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles
Sent: Monday, February 12, 2007 5:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] T1
On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote:
From what I read on the voip-info page, they did not define how to have
the phone muted when a paging call would go out. I already have the
paging (without mute) working using those same headers from that site.
But they don't cover the issue I'm
hello,
i ve a lot of problème with zaptel 1.4 when i tried to complile it
under debian kernel 2.4;
so i need to compile a new kernel version 2.6.x but id don't witch
kernel version is stable
van you help me please? :)
Younss AZ
KASTERISK.COM
skype: younssiga
On 2/12/07, Radu Padure [EMAIL PROTECTED] wrote:
I recommend you to use Sangoma A102D or A104D.
I agree, though if you are on a budget, the A101 + software echo
cancellation is pretty functional these days.
Cheers,
Steve.
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Well an upgrade to 1.2.17 now results in blips in the audio, instead of it
dropping. Guess it's time to go to SuperMicro.
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I'm working on writing some test IVR code in AGI. I can't get my FXO port to
detect a hang-up, but I'm going to deploying this using Digital cards so I
decided to just skip that problem for now. However this leaves me with a
question. How does AGI detect a hang-up if everything is operating
Steve,
I posed a similar question to Shane, but maybe you'll know as well..
I was able to get app_page to work. So when I call... **8050, it auto
answers and the callee is muted. However, what if that person wants to
answer the page and pickup to talk. They are already muted. Can you
unmute if
I whole heartedly agree.
Trixbox/FreePBX are much more mature and feature rich. AsteriskNOW has
greater future potential because of it's tight integration and no need for
MySQL/Apache. However, it's not there yet so if I was to implement
something today I would go with FreePBX. Not Trixbox but
Has anybody noted that under the latest release of Asterisk (1.2.15),
using the Features.conf for parking (I have ## setup) that it works fine
from SIP to SIP, but is sort of broken when trying it via IAX?
When parking via IAX with ##, you hear
'transfer'
Enter 700 for the parking extension
2007/2/12, younss azzayani [EMAIL PROTECTED]:
hello,
i ve a lot of problème with zaptel 1.4 when i tried to complile it
under debian kernel 2.4;
so i need to compile a new kernel version 2.6.x but id don't witch
kernel version is stable
van you help me please? :)
Younss AZ
KASTERISK.COM
skype:
On Mon, Feb 12, 2007 at 03:48:31PM +, younss azzayani wrote:
hello,
i ve a lot of problème with zaptel 1.4 when i tried to complile it
under debian kernel 2.4;
I didn't ;-)
so i need to compile a new kernel version 2.6.x but id don't witch
kernel version is stable
van you help me
Hello,
I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a
similar question: if someone is going to install Asterisk, FreePBX
and A2Billing, should you advice him/her to use Trixbox ... or a
custom step by step installation on a distribution of his/her choice?
Thanks
Stefano
On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote:
Steve,
I posed a similar question to Shane, but maybe you'll know as well..
I was able to get app_page to work. So when I call... **8050, it auto
answers and the callee is muted. However, what if that person wants to
answer the page and pickup to
Thank you, it's very easy than what i tink :)
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i've found a solutions working like this:
1 - I set up a queue that accepts caller even if it is empty.
2 - I set up an extension that dials an Agi script
3 - Each 5 seconds i run a cron job that:
- if the Agi script is busy (i parse the Action : Status) to detect
that : DO NOTHING
- if the Agi
Am Friday 02 February 2007 23:48 schrieb BJ Weschke:
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote:
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not
Hi,
I'd recommend if you need quick and easy setup - use [EMAIL PROTECTED] or
Trixbox
or something like this, and if you need customized setup and want to
understand system in detail - use your favorite distribution and setup *
from sources.
I'm prefer Slackware for any * installation, but your
Hi
We have a SER + asterisk server, on the same computer.
after starting sendmail service , the ser will be
confused.
we need sendmail to send voicemails .
best
Mani
Never Miss an Email
Stay connected with
Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it
actually posted.
The below worked for normal transfers. Now here is another situation. When we
try to transfer a call directly to voicemail it plays the voicemail message but
we can't transfer the call. The only
On Feb 8, 2007, at 6:55 AM, Tomislav Parčina wrote:
In article
[EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other sort of
It's all in the local LAN network - client computers (with SIP
softphones) are connected and registered at Asterisk SIP proxy via 100
MB connection each.
The QoS is enabled under TCP/IP protocol in LAN connection in Windows
(cause SIP softphones are running in Windows environment), and tos in
It's advisable to run it on any TDM FXO interface. Even with hardware
echo cancellation, fxotune makes it easier for the echo canceler to do
its job.
Matthew Fredrickson
On Feb 12, 2007, at 7:56 AM, Jerry Geis wrote:
Is it still needed to run fxotune on the TDM24XXE cards with
hardware
I am having trouble getting Asterisk to compile the zaptel stuff.
Here are the specifics:
Linux Kernel 2.5.9-42.0.8.EL
Asterisk 1.4.0
I compiled libpri, zaptel, asterisk and asterisk-addons (in that
order). This is a fresh install of CentOS. Following the CentOS
install, I did yum -y update
Stefano Corsi wrote:
Hello,
I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar
question: if someone is going to install Asterisk, FreePBX and
A2Billing, should you advice him/her to use Trixbox ... or a custom
step by step installation on a distribution of his/her
Mitch Thompson wrote:
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as
load
I have installed the vzaphfc and this what I was looking for.
I will do some more testing and I will post my results here.
Thank you for your help
Le 12 févr. 07 à 11:56, Tzafrir Cohen a écrit :
On Mon, Feb 12, 2007 at 09:15:01AM +0100, Olivier wrote:
There is currently a reimplementation
I currently have a customer that a previous employee setup with
Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom
menu, which I would assume is easily replicable in AsteriskNOW. The only
other thing I can think of is the sound bites for the menus. Does anyone
have any
The problem will be on the outside of your Asterisk PBX. In other words,
your asterisk server's external NIC (or if just one NIC), connection to your
firewall/router, to you voip provider.
You need to run tracert's from your Asterisk box to your voip provider.
QoS on the windows clients is
re Hi,
I m looking for a good document that allow me to install zaptel libpri
asterisk without errors, i ve a TDM400 TE110P, so please can you
help me
Kind Regards
Younss AZZAYANI
KASTERISK.COM
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i forgot to tell you that i m using a debian 2.6.8 kernel version
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Of course, you should take this with a grain of salt since I tried [EMAIL
PROTECTED]
(now TrixBox) for a total of 2 weeks before gutting it. Now, I just
use
my own GUI for everything from graphical setup to scripting.
There is nothing wrong with starting out with Trixbox. I still use
cOMO REALIZAS LA INSTALCION,
DEBES CONSEGUIR LOS TAR.GZ DE ZAPTEL y luego compilar asi:
#yum install kernel kernel-devel gcc
Con esto actualizas el kernel con sus fuentes y el gcc
#vi /etc/grub.conf
seleccionas el nuevo kernel como por default
# reboot
reinicias para que se ejecute el nuevo
I would suggest you grab the menu from the .conf file and paste it into the
new setup. (After even a little asterisk experience, they should be able to
get away from the gui).
The sound files could be copied as well. I'm guessing from your question
that you/your client may not having Linux
Hi Stefano,
I am a proponent of the step-by-step installation on a complete linux
distribution. Like someone said in another posting, the GUIs are nice, but
isolate you from the .conf files to the point where customization can be a
bit tricky. However, Trixbox w/ FreePBX and A2Billing works out
Have you tried this link?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debi
an.html
Edward Halman
(718) 705-7451
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, February 12,
Michael Collins wrote:
Of course, you should take this with a grain of salt since I tried [EMAIL
PROTECTED]
(now TrixBox) for a total of 2 weeks before gutting it. Now, I just
use
my own GUI for everything from graphical setup to scripting.
There is nothing wrong with starting out with
no, i'll try bouththe linke that you gave me the link that Cohen had
given to me
thank you very mutch Cohen Edward
2007/2/12, Edward Halman [EMAIL PROTECTED]:
Have you tried this link?
http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_debi
an.html
Edward Halman
(718)
in your dialplan:
[context]
...
h,1,AGI(...)
David Ruggles [EMAIL PROTECTED] wrote: I'm working on writing some test IVR
code in AGI. I can't get my FXO port to
detect a hang-up, but I'm going to deploying this using Digital cards so I
decided to just skip that problem for now. However this
I am curious how others handle call out VOIP and callerid. I have found
numerous providers that are cheap and seem to have good voice quality but
offer no provisions for callerid. I find it almost useless to use call
out when the receiving party gets some bogus callerid number that has no
That's right, but i think that you should use:
exten = h,1,DEADAGI( )
because in h extension the channel is considered as 'dead channel' ,
Regards,
--
J. Espinal
Slackware-es.com
chester c young wrote:
in your dialplan:
[context]
...
h,1,AGI(...)
*/David Ruggles [EMAIL PROTECTED]/*
I'm wondering if anyone else has experienced this. Up until a few days ago,
when accessing the CLI from my terminal program (Private Shell), the output
was in color. I haven't upgraded, rebuilt, or to my knowledge, changed
anything in Asterisk that would change this. My terminal settings were
On Mon, Feb 12, 2007 at 02:06:31PM -0500, Michelle Dupuis wrote:
I would suggest you grab the menu from the .conf file and paste it into the
new setup. (After even a little asterisk experience, they should be able to
get away from the gui).
Note that confiugration of AsteriskNow rewrites
That is incorrect. AsteriskNOW (actually, the AsteriskGUI) edits files in
place, leaving any old information in them. This allows you to fully customize
your users and dialplan without interfering with the GUI's operation.
Aaron Daniel
Community Relations Specialist
[EMAIL PROTECTED]
(256)
Lacy Moore - Aspendora wrote:
I'm wondering if anyone else has experienced this. Up until a few
days ago, when accessing the CLI from my terminal program (Private
Shell), the output was in color. I haven't upgraded, rebuilt, or to
my knowledge, changed anything in Asterisk that would change
Incorrect. I connect to asterisk -r all the time and get colour. Is it
possible your terminal emulation has changed in Private Shell? Is it VT100,
or ANSI?
On 2/12/07, Earle Clubb [EMAIL PROTECTED] wrote:
Lacy Moore - Aspendora wrote:
I'm wondering if anyone else has experienced this. Up
Hi Guys
I am just looking around for a small billing program but can't really find
what I am looking for.
It needs to bill straight off the CDR. It should grab all the CDR records
from the asteriskcdrdb mysql database then have a rates table to that it
calculate a bill from. Is there any
I have seen this when I have restarted the server from the asterisk CLI and
not a service asterisk restart command. I'm not sure as to why, but I always
assumed it had to do with the safe_asterisk file.
On 2/12/07, Earle Clubb [EMAIL PROTECTED] wrote:
Lacy Moore - Aspendora wrote:
I'm
On Mon, Feb 12, 2007 at 04:20:44PM -0500, Matt wrote:
Incorrect. I connect to asterisk -r all the time and get colour. Is it
possible your terminal emulation has changed in Private Shell? Is it VT100,
or ANSI?
Asterisk seems to disable the colors when you don't start it in a
terminal.
Apart from the feature/maturity issue there is a far more important
(IMHO) difference in the architectural approach of the two GUIs.
FreePBX assumes it owns the world and completely re-writes its
configuration files once changes are made through the GUI. While it
makes token efforts to enable
Aastra are a delight -- no need for a compiler (like the Grandstream and
Linksys phones) -- and extremely well documented configuration files.
--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
On Mon, 12 Feb 2007, George Pajari wrote:
Aastra are a delight -- no need for a compiler (like the Grandstream and
Linksys phones) -- and extremely well documented configuration files.
While I agree that Grandstream phones might not be the easiest things in
the world, I did find this
On 2/12/07, Bruce Reeves [EMAIL PROTECTED] wrote:
I have seen this when I have restarted the server from the asterisk CLI
and not a service asterisk restart command. I'm not sure as to why, but I
always assumed it had to do with the safe_asterisk file.
Bruce, that may have been it. I just
On Mon, Feb 12, 2007 at 11:27:48PM +0200, Tzafrir Cohen wrote:
On Mon, Feb 12, 2007 at 04:20:44PM -0500, Matt wrote:
Incorrect. I connect to asterisk -r all the time and get colour. Is it
possible your terminal emulation has changed in Private Shell? Is it VT100,
or ANSI?
Asterisk
Does anyone know how I could get the SayUnixTime application to say
files from a different sound directory?
It looks like it uses the language as a base to determine where to play
sound files from. I need to override that.
Thanks,
Doug.
___
On Mon, Feb 12, 2007 at 04:30:54PM -0600, Lacy Moore - Aspendora wrote:
On 2/12/07, Bruce Reeves [EMAIL PROTECTED] wrote:
I have seen this when I have restarted the server from the asterisk CLI
and not a service asterisk restart command. I'm not sure as to why, but I
always assumed it had to
Matthew,
ok, but is realy possible change the dsp code in the Asterisk? Guys around
The OpenPBX change the dsp to Steve's spandsp and has the native T38 support
now.
Tomas Urbanek
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
On Mon, 2007-02-12 at 11:59 -0600, Butch Evans wrote:
I am having trouble getting Asterisk to compile the zaptel stuff.
Here are the specifics:
Linux Kernel 2.5.9-42.0.8.EL
Asterisk 1.4.0
I compiled libpri, zaptel, asterisk and asterisk-addons (in that
order). This is a fresh install of
Well an upgrade to 1.2.17 now results in blips in the audio, instead of it
dropping. Guess it's time to go to SuperMicro.
1.2.17 ? (1.2.13 zaptel?)
i have supermicro mobo(P8SCT) and have same problem with shared
interrupts
bash#lspci -bv | grep -i IRQ 5 --before-context=2
00:02.0 VGA
Hi Mark,
Take a look at the YakaVOIP solution from http://www.yakasoftware.com/
http://www.yakasoftware.com. Probably suits your requirements.
Greetz,
Roland.
_
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von MBIT
Technologies
Gesendet: 12 February 2007 22:23
On Mon, Feb 12, 2007 at 11:59:55AM -0600, Butch Evans wrote:
I am having trouble getting Asterisk to compile the zaptel stuff.
Here are the specifics:
Linux Kernel 2.5.9-42.0.8.EL
Asterisk 1.4.0
I compiled libpri, zaptel, asterisk and asterisk-addons (in that
order). This is a fresh
Noc Phibee wrote:
Hi
i have a big problems with my asterisk .. i use a Digium TDM400P for
connect a
analog line.
And not all time (i don't know why) he don't see the end of the call and
anyone can call me
(occuped)
For that's work, i am disconnect the phone cable and it's good
Mohamed Farid wrote:
On 2/12/07, Mohamed Farid [EMAIL PROTECTED] wrote:
I have a Cisco Call Manager - and need to use the IVR Feature from
Asterisk.
My extension is 400 and I am calling 558 on Asterisk
In my extension.conf I have these lines :
exten = 558,1,Answer
exten =
Michelle Dupuis wrote:
The problem will be on the outside of your Asterisk PBX. In other words,
your asterisk server's external NIC (or if just one NIC), connection to your
firewall/router, to you voip provider.
You need to run tracert's from your Asterisk box to your voip provider.
Lee Jenkins wrote:
Stefano Corsi wrote:
Hello,
I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a
similar question: if someone is going to install Asterisk, FreePBX and
A2Billing, should you advice him/her to use Trixbox ... or a custom
step by step installation on a
Michael Collins wrote:
Of course, you should take this with a grain of salt since I tried [EMAIL
PROTECTED]
(now TrixBox) for a total of 2 weeks before gutting it. Now, I just
use
my own GUI for everything from graphical setup to scripting.
There is nothing wrong with starting out
Er... no you don't :)My problem and everyone elses with Dell is that
Dell builds the Mobos to share the PCI IRQs with the NIC cards. I've got
some SuperMicro MoBos running VoIP and they DO share exactly like you
showed.
There is nothing wrong with sharing your VGA (Video) with your PSTN
ARG. I need to stop posting to this list until I recover from this cold.
I see now that you are sharing with your NIC card. That *IS* bad. Go into
the BIOS. Turn off USB, and Parallel, Serial... basically everything you
don't need. Now, save and reboot. Go back into BIOS. See if you can,
IMHO,
If you don't know enough about Linux/Asterisk/FreePBX to be able to set it
up yourself you should not be doing it for a Production install in a
business environment.
NOTE: Production install in a business environment does NOT include setting
it up in your house with extensions for the kids
If you are installing in a business environment I think being able to use
both and know the benefits of both are pretty essential.
FreePBX is a great tool and should be used to its potential because it has
some great features. It can also lessen the time it takes to do an install.
Some have said
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