Hi All,
Apologies for this, migrated the site and forgot to change a path. Site's
back up now.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Sunday, July 27,
A customer has an Asterisk box with two TDM400P cards, running [EMAIL
PROTECTED] 2.8,
which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
for a while, with only some small issues. They have FXO ports going
to analogue POTS (UK standard) lines, and SIP phones for extensions.
I
I have a Digium Appliance AA50 configure with 8 lines and two dial plans.
Each dial plan takes care of a particular location. In Dialplan2 we have 4
lines.
xxx xxx 0333
xxx xxx 0005
xxx xxx 0006
xxx xxx 0007
When a call gets to line 0005 you pick up the phone but the call does not
get
Olivier wrote:
tftp is the client, do you have it installed ?... example:
# tftp hostname
tftp get /srv/tftp/foo.txt
tftp ^D
# cat foo.txt
...
That was exactly what I was after : I installed tftp on my Ubuntu
system and checked Debian tftp server
I have just received a list of requests from one of our customers and I
really do not have the time or knowledge to work on it.
I will truly appreciate it if someone could help me outside of the mailing
list. Please contact me if interested.
Digium Appliance AA50/GrandStream GXP2000.
Here
Have you tried to contact Digium Support about this question?
They are very good at answering AA50 configuration questions.
Regards,
Doug Bailey
- Fidel Garcia [EMAIL PROTECTED] wrote:
I have a Digium Appliance AA50 configure with 8 lines and two dial
plans. Each dial plan takes care
On Sat, Jul 26, 2008 at 06:33:35PM -0400, Eric ManxPower Wieling wrote:
The something is generated by Asterisk at the time the call is
created. You should never add it, since you don't control that call
instance info. In fact, you should almost never care about the call
instance string.
Hey,
I have a few dids with callcentric. They seem to work fine most of the
time but at some points I get handle_request_invite: Failed
to authenticate user sip:PSTNnumber
This happens intermittently.
The way I understand it the insecure=port,invite should tell asterisk
not to authenticate
emist wrote:
Hey,
I have a few dids with callcentric. They seem to work fine most of the
time but at some points I get handle_request_invite: Failed
to authenticate user sip:PSTNnumber
This happens intermittently.
The way I understand it the insecure=port,invite should tell asterisk
Hello list,
I want to use Asterisk as a PBX connected to a public SIP service
provider as uplink.
The environment where I want to deploy the solution makes it necessary
to request (IP guaranteed quality of service) resources per active call.
This is why I am looking for a way to interface a
Hi list:
Is there any way, to set a common inter-queues leastrecent Strategy, i'm
searching a Behaviour like this:
2 Queues Q1 and Q2
2 Agentes A1 and A2
Both agents are in both queues.
First Call in the system is for Q1 and is answer by
Hi guys, I know this problem has just been fixed in trunk
(http://bugs.digium.com/view.php?id=7403), but I'm asking for a
workaround for previous versions of Asterisk, as we can't run off of
Trunk (e.g. we have to run with Asterisk from Ubuntu 8.10).
Basically, I have a situation where I have
At 01:58 AM 7/28/2008, you wrote:
Does anyone know if there have been changes in the 1.2 series that
affects ring detection on the TDM400P FXO ports? Any critical settings
in zaptel.conf or zapata.conf?
Some of us know there were changes because we've experienced the same failure.
Ira
Quote
Recently I discovered a cool new site called Google.
They have lots of information about ISDN cards. :-P
Grüße,
Philipp Kempgen
Yes - There is also a lot of bogus, incorrect, crap.
His question was fair, on-topic, politely asked and as such hardly deserves to
be
made fun off
Dean
Robby Dermody wrote:
Hi guys, I know this problem has just been fixed in trunk
(http://bugs.digium.com/view.php?id=7403), but I'm asking for a
workaround for previous versions of Asterisk, as we can't run off of
Trunk (e.g. we have to run with Asterisk from Ubuntu 8.10).
Basically, I have a
On Mon, Jul 28, 2008 at 10:10:49AM -0700, Ira wrote:
At 01:58 AM 7/28/2008, you wrote:
Does anyone know if there have been changes in the 1.2 series that
affects ring detection on the TDM400P FXO ports? Any critical settings
in zaptel.conf or zapata.conf?
Some of us know there were
I really don't see any issue with running it on a laptop, you even
have a built in UPS. That is of course if you need to be up with some
guarantee. I have first gen Pentium based Dell Latitudes with several
years of uptime.
Anyways, with two laptops configured identically and this
Tony Mountifield wrote:
A customer has an Asterisk box with two TDM400P cards, running [EMAIL
PROTECTED] 2.8,
which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
for a while, with only some small issues. They have FXO ports going
to analogue POTS (UK standard) lines, and
Hi,
Would just like to know if it's possible to be able to call a macro at the same
time.
i use a macro to dial local extension to another extension.
exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)
but now i would like to use it on a simple ringgroup where it will
OS: CentOS 5.2
Asterisk: 1.4
I use NeoSpeech to do TTS recording. When I play back the files in
Asterisk, the playback seems to be at about half-speed or less.
However, when I play them through totem, xmms or other audio
applications they sound fine. I've tried recording in 8-bit Mu-law PCM
You could pick up a couple of these cute little guys
http://www.surpluscomputers.com/store/Main.aspx?p=ItemDetailitem=com10791
Make them identical and then use the single Redfone PRI box
http://www.voipsupply.com/product_info.php?products_id=4069osCsid=3ea19c2324c85f2923deb8dfd25c2cf4
That is
Drew Gibson wrote:
Tony Mountifield wrote:
A customer has an Asterisk box with two TDM400P cards, running [EMAIL
PROTECTED] 2.8,
which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
for a while, with only some small issues. They have FXO ports going
to analogue POTS (UK
Hey Dave,
Thanks for the help. Thats about the only thing I didn't think to try.
It seems to have resolved the problem.
Regards,
Igor H.
Dave Fullerton wrote:
emist wrote:
Hey,
I have a few dids with callcentric. They seem to work fine most of the
time but at some points I get
In article [EMAIL PROTECTED], Drew Gibson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
A customer has an Asterisk box with two TDM400P cards, running [EMAIL
PROTECTED] 2.8,
which has Asterisk 1.2.7.1 and Zaptel 1.2.5. This has been running ok
for a while, with only some small issues.
Tony Mountifield wrote:
My guess is that 1.2.24 will work, which was at revision 3842 of the tree
(rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks innocuous
enough, but the follwing two (revs 4128 and 4132) look likely culprits,
from looking at the areas of code that they
you can try to place your macro extensions into single dialgroup using
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})
ronald ramos wrote:
Hi,
Would just like to know
hi,
thanks for your reply. is dialgroup already available in asterisk 1.4?
i'm currently using 1.4.21.
regards,
ron
--- On Mon, 7/28/08, Pavel Jezek [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users] simultaneous dial macro
To: [EMAIL PROTECTED],
New in Asterisk 1.6
ronald ramos wrote:
hi,
thanks for your reply. is dialgroup already available in asterisk 1.4?
i'm currently using 1.4.21.
regards,
ron
--- On Mon, 7/28/08, Pavel Jezek [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users]
Hi,
How can i enable the if you know your parties extensions please dial it
now function? what do i need to add below?
[ivr-1]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(custom/myivr)
exten =
Does anyone have any suggestions on what to use to monitor a vendor doing
remote support?
On the windows side things are typically done via screen sharing (
gotoassist.com, bomgar or similar) so at least you can see what the other
end is doing.
In working with linux (especially hardware vendors
Take a look at 'screen'. Chances are, it's already installed on your boxen.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Original Message -
From: Joe Pukepail [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am testing the imap voicemail funtionality. I compiles asterisk using
version 1.4.21.2 on rhel5.1.
I have two different customers provisioned on the same asterisk as follows:
taken from voicemail.conf:
imapserver=192.168.196.43
imapflags=notls
authuser=asterisk
authpassword=asterisk
;
Al Baker schrieb:
Quote
Recently I discovered a cool new site called Google.
They have lots of information about ISDN cards. :-P
Grüße,
Philipp Kempgen
Yes - There is also a lot of bogus, incorrect, crap.
His question was fair, on-topic, politely asked and as such hardly deserves
Nhadie wrote:
Hi,
How can i enable the if you know your parties extensions please dial it
now function? what do i need to add below?
[ivr-1]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten =
Joe Pukepail schrieb:
Does anyone have any suggestions on what to use to monitor a vendor doing
remote support?
On the windows side things are typically done via screen sharing (
gotoassist.com, bomgar or similar) so at least you can see what the other
end is doing.
In working with linux
Philipp Kempgen wrote:
I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ).
screen doesn't solve the security aspect of your question though.
Grüße,
Philipp Kempgen
Actually, it could. What I've done before, is give out an unprivileged account
on the box (or some
Deric Page wrote:
OS: CentOS 5.2
Asterisk: 1.4
I use NeoSpeech to do TTS recording. When I play back the files in
Asterisk, the playback seems to be at about half-speed or less.
However, when I play them through totem, xmms or other audio
applications they sound fine. I’ve tried
On Jul 28, 2008, at 5:50 PM, Jason Parker wrote:
Philipp Kempgen wrote:
I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ).
screen doesn't solve the security aspect of your question though.
Grüße,
Philipp Kempgen
Actually, it could. What I've done before, is give out an
This time, I am trying to remotely install Asterisk in China.
I was told that an E1 line has been installed and so I plug it into port
1 of a TE412P.
On the box, first of all, I just installed Zaptel 1.4.10.1.
# service zaptel restart
Unloading zaptel hardware drivers:ERROR: Module zaptel is in
My best guess from looking at that is that its a driver bug. The last
thing that happens before the lockup seems to be an ioctl call to the
device.
Hope it helps,
Igor H.
Lee, John (Sydney) wrote:
This time, I am trying to remotely install Asterisk in China.
I was told that an E1 line has
My best guess from looking at that is that its a driver bug. The last
thing that happens before the lockup seems to be an ioctl call to the
device.
Hope it helps,
Igor H.
Thanks Igor.
Does it mean that I should install a later release of zaptel?
On Monday 28 July 2008 22:48:26 Lee, John (Sydney) wrote:
This time, I am trying to remotely install Asterisk in China.
I was told that an E1 line has been installed and so I plug it into port
1 of a TE412P.
Are you sure that they're plugged into port 1 and not port 4? It is a rather
common
Hello,
I am having an issue here that after an attended call transfer there is no
audio on one way; the problem is caused by Asterisk sending two INVITE messages
without waiting for an ack for the first one.
The issue has been reported on bug 9305, has been fixed and the fix is now
included
Are you sure that they're plugged into port 1 and not port 4? It is a
rather
common mistake to believe that the port numbers start at the bottom of
the card and not at the top.
Thanks Tilghman.
I checked with the guys in the remote office and he is certain that he
has plugged the E1 line
I think it can't hurt to try a different release. Let me know how it goes.
Regards,
Igor H.
Lee, John (Sydney) wrote:
My best guess from looking at that is that its a driver bug. The last
thing that happens before the lockup seems to be an ioctl call to the
device.
Hope it helps,
Igor H.
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