[EMAIL PROTECTED] is believed to have said:
The error is the same, afaik.
What I can't understand is why the make is entering in the directory '/
usr/src/linux-2.6.11.4-21.7-obj/i386/default'; I am by far not expert,
but I would expect it to go fiddle with a '586' directory.
Just a
[EMAIL PROTECTED] is believed to have said:
and watch linus himself rant about how this is incorrect to do (yet all
the distros do it) :P
Well, this is reassuring for a newbie like me.
Even the pros (as anybody building a distro ought to be, and most of the
times, really is) can do obvious
On Thu, 2005-21-07 at 23:08 -0500, Steve Maroney wrote:
I have the original version of the IAXY. I had it laying around collecting
dust, now Im actually putting it to use. When I call my voicemail
extension (8500), Before I get the voice prompts from the voicemail app,
I hear tones that sound
I have a queue setup using Asterisk CVS and roundrobin, however calls
seem to be distributed in the same way as rrmemory (round robin with
memory), ie, it is alternating between the two people in the queue
rather than always calling the first available person in the queue
first.
I am using agents
Juraj Bednar wrote:
Hello,
There's some work on creating a multiprotocol solution for instant
messaging within Asterisk, but it will not be in the coming v1.2.
is the work somewhere as a patch to be tried or in some other form,
even if it's not coming to 1.2?
No, there needs to be
Hi,
I installed a Quadbri card, configured
with 2 ports connected to a Hipath pbx an 2 ports connected to telco.
I can make and receive call but I receive
every 5 seconds on asterisk cli the message: No D-channel available
Using Primary on channel aniway 12
Primary D-channel on span 4 up
My
Hi all
Xorcom Rapid 1.1 is here.
* Asterisk 1.0.9
* Flash Operator Panel
* improved Zaptel hardware detection: should hopefully detect E1, T1
ZapHFC and qozap. No more channel numbers guessing in
zapata/zaptel.conf
* and much of the extra software available for it
You can get the full
On Fri, 2005-07-22 at 07:59 +0200, Aldo Bergamini wrote:
[EMAIL PROTECTED] is believed to have said:
and watch linus himself rant about how this is incorrect to do (yet all
the distros do it) :P
Well, this is reassuring for a newbie like me.
Even the pros (as anybody building a
It's not a spam. They are not yokels. Don't know about you
Gizmo is basically a different front end offered by the Sipphone.com people, to
offer an alternative to Skype which is not a closed jail (interoperates with all
SIP devices, asterisk, etc.).
I think they sent the mail to all
On Thu, Jul 21, 2005 at 04:52:43AM -0700, trixter http://www.0xdecafbad.com
wrote:
On Thu, 2005-07-21 at 07:36 -0400, Doug Lytle wrote:
Just a guess, your simlink is pointing to the incorrect linux source
directory. Go into /usr/src/linux and do a ls -l, see where the linux
simlink is
On Thu, Jul 21, 2005 at 09:45:02PM +0100, Angus Comber wrote:
I am now getting this make error:
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
Remove the line that
On a different note using Fedora Core 3 I get
CC [M] /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c: In function `zt_chan_write':
/usr/src/zaptel/zaptel.c:1745: warning: ignoring return value of
`copy_from_user', declared with attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In
On Fri, 2005-07-22 at 08:31 +0100, Lee Archer wrote:
On a different note using Fedora Core 3 I get
When building the stable or head zaptel with kernel linux-2.6.11-1.35_FC3.
The module compiles but it never used to give this message on FC2.
Anyone got any ideas?
compiler upgrade
Hi all, i've installed AMP and Asterisk following the INSTALL file and i have
a problem with the TDM04B with 4 FXO:
[EMAIL PROTECTED] ~]# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default)
I see the GXP2000 has a headset socket. Are their any compatible
headsets for it. How does the functionality change?
What else would people suggest for a Call-Centre?
Would like Headset, Call Details - etc...
The call centre answers the phone according to which number is called..
--
. .
Hi,
please post you zaptel.conf
Giorgio
Michele O-Zone Pinassi wrote:
Hi all, i've installed AMP and Asterisk following the INSTALL file and i have
a problem with the TDM04B with 4 FXO:
[EMAIL PROTECTED] ~]# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01:
You can traversal a HTTPS proxy using a plain TCP connection (without SSL).
The unique requirement of some HTTPS proxys is that the target port is 443.
Then if your Asterisk listen in 443 port IAX (TCP) connections, it should
work.
G.
-Mensaje original-
De: [EMAIL PROTECTED]
Title: SATA
Has anyone had any problems with SATA, either on board or 3rd party setup? I've currently got a problem where an AMD non SATA FC2 system is working fine but an Intel system with a 3Ware SATA card and FC3 is radomly not syncing with the ISDN30. It allows and receives calls but at
Hi,
if you use version 3 you cannot go back to previous version without
sending your phone to Polycom.
If you want we have some files but we do not guarantee the right working.
Which version do you need?
Giorgio
Michael Felder wrote:
Hello,
Does anybody have the latest Boot ROMs for the
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this msg:
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??,
thx in advance,
CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de 1,49
Hi,
try to add the gateway field in your config file...
My conf file:
ip: 192.168.100.2
netmask: 255.255.255.0
gateway: 192.168.100.3
codec: ulaw
server: 192.168.100.3
user: iaxy
pass: iaxy
register
Giorgio
Bryce Chidester wrote:
On Wed, 2005-07-20 at 08:26 -0400, Ousmane Doukara wrote:
Hello, dear Asterisk experts.
When I run Asterisk (CVS HEAD version), I'm not able to play music
anymore -- asterisk seems to capture sound device. Is it not a bug,
but a feature? That's unlike stable (1.0.7 and 1.0.9) versions, when
I can, say, run an IP telephone on the *same* machine and
On Friday 22 July 2005 11:37, Giorgio Incantalupo wrote:
Hi,
please post you zaptel.conf
Giorgio
Thanks i've solved: i'm using a wrong modules combination :-D
Thanks however ! Oz
--
O-Zone ! No (C) 2005
www.zerozone.it
___
Asterisk-Users
How strange - that worked! I wonder why that was put there?
Angus
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, July 22, 2005 8:29 AM
Subject: Re: [Asterisk-Users] Problems installing asterisk-addons
On Thu, Jul 21,
Hey!
My asterisk is working properly so far with all automatic functions. Now I want
to direct incoming calls to operator, i mean some person who answers to the
incoming calls and redirect them to the person caller wants.
What I have so far searched from the voip-info.org and other sites, Ihave
Hi,
i've just faced with some bridged calls which could not be hungup just
killing the asterisk process solved the problem:
Zap/63-1 (incoming s1 ) Up Bridged Call SIP/2035-e9cb
logs say:
Jul 22 14:54:12 NOTICE[17161] chan_sip.c: Disconnecting call
'SIP/2035-e9cb' for
Dear All,
I have a problem with the Marco and the Realtime Extensions in my
extensions.conf. The problem is that when I exit from my Marco, I
should return to my calling context, which is default but the next
step for it should be switch statement which will use realtime
extension. Somehow I am
Noah Miller wrote:
snip
In addition to largely being a rehash of existing docs on the internet,
there are many editorial errors in the version that I have. Before I
was comfortable with the conf files, these editorial errors were very
confusing. The editions coming out now may have fixed
hi all,
when i start * all zap channels get ring signal so i get a huge number
of incoming dummy calls when starting *. i'm using TE105P with 4 TA750
full fxo with latest CVS HEAD:
zaptel.conf:
span=1,0,0,esf,b8zs
fxsks=1-24
span=2,0,0,esf,b8zs
fxsks=25-48
span=3,0,0,esf,b8zs
fxsks=49-72
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this msg:
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??,
thx in advance,
CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de
On Fri, Jul 22, 2005 at 02:17:48PM +0400, Timur V. Elzhov wrote:
When I run Asterisk (CVS HEAD version), I'm not able to play music
anymore -- asterisk seems to capture sound device. Is it not a bug,
but a feature? That's unlike stable (1.0.7 and 1.0.9) versions, when
I can, say, run an IP
hi Tzafrir,
i was able to run make by removing ^M at the end of each line of each
script, i also checked all script file on the /asterisk folder and execute
dos2unix command on all script files, however when i run make i encountered
another problem.
gcc -pipe -Wall -Wstrict-prototypes
Round robin is designed to alternate between, in this case, the two agents.
At least that is how I understand the comment in the queues.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Thursday, July 21, 2005 11:18 PM
To:
On Fri, Jul 22, 2005 at 11:39:04AM +0200, Gustavo García wrote:
You can traversal a HTTPS proxy using a plain TCP connection (without SSL).
The unique requirement of some HTTPS proxys is that the target port is 443.
Then if your Asterisk listen in 443 port IAX (TCP) connections, it should
David Romero a écrit :
attended transfer are implemented on some cases on the phone side, if
you need attended transfers on dial plan you need use asterisk CVS
HEAD, i are using asterisk CVS HEAD and attended transfer work very well.
just install asterisk CVS HEAD and configure features.conf
Hi,
don't think it is the cable length because we tried to shorten it and
nothing changed. We tried to use a Dell Poweredge with a TDM400P and a
quadBRI using bristuffed-Asterisk 1.0.7 with no success...the only
solution was removing tdm400P.
We checked the interrupts but the two cards had
Hello,I am looking for 4 port FXO low profile PCI
card that could be used with Asterisk.Digium TDM04B sound like a good
choice but it is half high PCI card and I can not plug it in my Dell box (small
box).I am looking for adequate low profile PCI card (55mm high or similar
but definitely
Hi all,
chan someone who has tried BOTH chan_capi and chan_mISDN with a
passive Frtiz!Card PCI comment on one versus the other. Which had
better sound quality.
Thanks
___
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Asterisk-Users@lists.digium.com
Doing IAX over TCP is simply a Bad Idea.
Under perfect circumstances, it will work OK, but the slightest network
disturbance will result in sound gaps/distortion and/or monster audio delay.
This is not idle UDP-boosting, I've tried it.
[Have had good results with UDP-based secure tunnel
[EMAIL PROTECTED] wrote:
Hey!
My asterisk is working properly so far with all automatic functions. Now I want
to direct incoming calls to operator, i mean some person who answers to the
[incoming]
exten s,1,Answer()
; Wait for 2 seconds to pick up caller-id
exten s,2,Wait(2)
; Dial
Hi,
the question is: can digium and quadBRI co-exists easily on the same server?
We are still having a lot of troubles since it is hard to find infos on
how to configure them.
Giorgio.
Emanuele Pucciarelli wrote:
Kevin Hanson wrote:
Can anyone recommend a BRI card that supports
youa re using -v option multiple times at startup.
That message is perfectly fine.
ali kia wrote:
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this
msg :
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2
(Ring/Answered)...
someone
On Fri, Jul 22, 2005 at 01:43:01PM +0300, [EMAIL PROTECTED] wrote:
Hey!
My asterisk is working properly so far with all automatic functions. Now I
want
to direct incoming calls to operator, i mean some person who answers to the
incoming calls and redirect them to the person caller wants.
David Romero a crit:
attended transfer are implemented on some cases on the
phone side, if
you need attended transfers on dial plan you need use asterisk CVS
HEAD, i are using asterisk CVS HEAD and attended transfer work very
well.
just install asterisk CVS HEAD and configure features.conf
sylvain garcia a crit:
David Romero a crit :
attended transfer are implemented on some cases on the phone side, if
you need attended transfers on dial plan you need use asterisk CVS
HEAD, i are using asterisk CVS HEAD and attended transfer work very well.
just install asterisk
Hi Asterisk-Users,
We have a problem with queues.conf / extensions.conf
queues.conf file reads like ...
member = SIP/8399
extensions.conf reads like ...
exten = 8399, 1, SetCIDNum(${AccountNumber}|a)
exten = 8399, 2, Dial(SIP/8399,10,Ttrf)
When somebody calls to the queue, we observed that
Empire Records - Money (that's what I want)
Erm, do you mean The Flying Lizards? My cousin was the vocalist on that
one hit wonder. She makes a fotune out of XM Satellite Radio. They play
it all the time on their 80's channel.
Julien Goodwin wrote:
On 22/07/2005 1:58 PM, Mark Phillips
Hi Folks,
I have an IAX trunk link to a collegues house. I'm using AAH and he's
got the latest CVS as of last Tuesday.
Problem we're having is this; when I dial his extension 7201 (Pulver
WiSIP phone) his * box sends me 1 ring and then Alison's busy message.
If I call his 7202 extension
On 22/07/2005 1:58 PM, Mark Phillips wrote:
Does anyone have a collection of stupid hold music? Y'know, the sort of
thing that would drive a person mad? Silly songs, repetative tunes etc?
My two (s)cents...
* Anything from New Kids on The Block.
* Put the pop-hook guitar riff from the
Sunshine, lolly-pops, and rainbows; everything that's wonderful is what I
feel when we're together ...
Thomas Christie
There are 10 types of people in the world: those who understand binary and
those who don't.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have an IAX trunk link to a collegues house. I'm using AAH and he's
got the latest CVS as of last Tuesday.
Problem we're having is this; when I dial his extension 7201 (Pulver
WiSIP phone) his * box sends me 1 ring and then Alison's busy message.
If I call his 7202 extension (X-Ten Pro
I
got the same problem, and what i change it was point-to-point
to point-to-multi-point, and I used normal patch cable to connect the ISDN card
to ISDN connector, and I used TE mode.
In
italy work like
that
I
hope that will help you,
I
used also bristuff.
Good
luck
Da:
Oh man, they are just plain B, not even worth viewing which is a
shame because this mp3 site looked promising.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jeremy Melanson
Sent: Friday, 22 July 2005 9:08 AM
To: Asterisk
I found the same problem, u have to use normal patch cable to connect from ISDN
card to ISDN connector,
And u r in italy I think, in italy u have to use p2mp TE mode
signalling=bri_cpe_ptmp, with telcom,
Good luck
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Hi Patrick,
Removing spaces didn't help in this regard.
Some other solutions?
Best regards
Somesh SS
--- Patrick [EMAIL PROTECTED] wrote:
On Fri, 2005-07-22 at 05:38 -0700, somesh s wrote:
exten = 8399, 1, SetCIDNum(${AccountNumber}|a)
exten = 8399, 2, Dial(SIP/8399,10,Ttrf)
Not that it need any additional 'push' against it, :-)..
My tests with IAX over OPENVPN (on port 443) are acceptable (they do
work just fine) for basic non-user-friendly purposes.
Examples, I get my voice mail at home sometimes via this tunnel (if wife
using primary landline.
I test my
Guys.
What do you think about Dell hardware and Asterisk? Whos using it, comments,
any special specs recommended or models?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
We are using this combination.
we are thinking about change the DELL computers!
Bruno De Luca Graziosi
Anton Krall wrote:
Guys.
What do you think about Dell hardware and Asterisk? Whos using it, comments,
any special specs recommended or models?
Happy Tree Friends' theme is all you need to annoy who's on hold
(even if actually I don't know if you can use it for business purpose)
Anyway, isn't time to split this list in strictly technical
questions-asterisk-users and what's the best
I have a small government department that wants me to implement a
Asterisk installation, however, they connect to the Government PBX, a
Mitel SX200, and want to keep the ability to do that. I know there is no
chance to connect the digital extension lines, but would it be possible
to have the
I am observing.
The problem is in the outbound calls.
Some are not completed.
Thank you
- Original Message -
From: Steve Totaro
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, July 21, 2005 11:33 PM
Subject: Re: [Asterisk-Users] T1 - incomplete calls
On Friday 22 July 2005 14:48, Anton Krall wrote:
Guys.
What do you think about Dell hardware and Asterisk? Whos using it,
comments, any special specs recommended or models?
http://www.digium.com/index.php?menu=compatibility
Digium's recommendation is quite clear: 'Don't use Dell hardware'
Il giorno ven, 22/07/2005 alle 08.48 -0500, Anton Krall ha scritto:
Guys.
What do you think about Dell hardware and Asterisk? Whos using it, comments,
any special specs recommended or models?
I'm using a DELL PE750 server with an AVM c2, suse-pro installed, capi
works out of the box. DELL
To All;
My current issues is a 5 second delay for call that is being transferred
from the Norstar units to
the Asterisk servers VIA a PRI. Is their anything that can be done to
speed up the transfer on the Norstar. Below is my current phone
config.
Norstar1 PRI Asterisk-1
Hi there,
Our problem is with outgoing calls...
And the problem is some calls do not complete...the asterisk show the
ring...but doesnt complete some calls...we dont have dropped calls...
thank you
- Original Message -
From: Paul Belanger [EMAIL PROTECTED]
To: Asterisk Users Mailing
YAACID 0.91 has been released. You can access it on the web site
http://www.shatterit.com/opensource/yaacid
this should fix some problems with [EMAIL PROTECTED] and older cvs and non-cvs
asterisk versions. (the manager interface has changed quite a bit,
which was causing the problems)
Theres
Boris Zolotarev - Pamet wrote:
Digium TDM04B sound like a good choice but it is half high PCI card and I can
not plug it in my Dell box (small box).
I am looking for adequate low profile PCI card (55mm high or similar but
definitely smaller than TDM04B so I can plug it in).
You will not
DELL computers ussualy has got IRQ conflicts with the USB and slots PCI.
If you disable the USB controller from BIOS you get a perfect server.
I have tried several PowerEdge 2850 like Asterisk dedicated server and
it's running perfectly.
I have tried IBM xServer 226 and 346 and the IRQ
Simone Cittadini wrote:
snip
Anyway, isn't time to split this list in strictly technical
questions-asterisk-users and what's the best
provider/hardware/moh/book/distro/Iwanttocomplaincostheydidn'tsendmethecdrom-asterisk-users
lists ?
Simone Cittadini
IT Manager
And lest we forget, another
[EMAIL PROTECTED] wrote:
And it's a great shame Digium hardware has such problems on
Dell kit, since
there's so much of it about :(
If you don't use digium hardware, there's of course no problems with using Dell.
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 3
There are 10 types of people in the world: those who understand binary and
those who don't.
why are these stupid quotes so amuzing?
love it!!On 22/07/05, Thomas Christie [EMAIL PROTECTED] wrote:
Sunshine, lolly-pops, and rainbows; everything that's wonderful is what Ifeel when we're together
pri show span 1
Primary D-channel: 24Status: Provisioned, Up,
ActiveSwitchtype: National ISDNType: CPEWindow Length:
0/7Sentrej: 0SolicitFbit: 0Retrans: 0Busy: 0Overlap Dial:
0T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer:
4000T313 Timer: 4000N200 Counter: 3
thks
-
My debugThank you for help.
Verbosity is at least 5 --
Accepting AUTHENTICATED call from
requested format = g729, requested
prefs = (), actual format =
gsm, host prefs =
(gsm), priority =
mine -- Executing AbsoluteTimeout("IAX2/[EMAIL PROTECTED]", "3600") in new
stack -- Set
On Friday 22 July 2005 10:15, Michael Di Martino wrote:
My current issues is a 5 second delay for call that is being transferred
from the Norstar units to
the Asterisk servers VIA a PRI. Is their anything that can be done to
speed up the transfer on the Norstar. Below is my current phone
Hi
I have my new TDM400P installed and working. I'm
running from cvs HEAD with a 2.6.12 kernel on debian.
I can't seem to get Caller id working (in uk with
clid supplied and working to line) but am a bit unclear on the docs and hence
assume it is something I am doing wrong.
I would
It would be helpful to capture a complete ISDN call setup.
On the cli type pri debug span 1
Then place a call and turn off debug with pri no debug span 1
You will then have a complete listing of the signalling between your
co and your * for this time period.
Good Luck
On Jul 22, 2005, at
I see the Dell SC420 is discarded according to Digium but what about the
SC430, SC1420 or others?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Simone Cittadini
|Sent: Viernes, 22 de Julio de 2005 09:12 a.m.
|To: Asterisk Users Mailing List -
I was thing about XTCs stupidly happy
M.
On Fri, 22 Jul 2005 15:57:07 +0200, Simone Cittadini wrote:
Happy Tree Friends' theme is all you need to annoy who's on hold
(even if actually I don't know if you can use it for business purpose)
Anyway, isn't time to split this list in strictly
Was this an exmple of your incomplete calls?
From the trace it appears that you issued the disconnect while the
call was in process.
On Jul 22, 2005, at 9:32 AM, JOAO CARLOS MOURA wrote:
My debug
Thank you for help.
Verbosity is at least 5
-- Accepting AUTHENTICATED call from
Hi,
I tried to install a tdm400P and a monoBRI. I loaded zaphfc and wcfxs
modules and everything seemed allright but linux log shows the following
message:
zaphfc: sync lost, pci performance too low. you might have some cpu
throtteling enabled.
Anybody knows what it means?
TIA Giorgio
sorry i have 1.0.7 version it's possible of attended transfer?
No, it is only available in CVS.
Udo
PS: Please don't quote all of the other messges you are replying to.
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Anyway, isn't time to split this list in strictly technical...
And lest we forget, another split for the Cisco/Polycom/Snom/othersipphone
configurations...
That would be short-sighted imo. Splitting the list or goofy offers to do
the list as a PHPbb, NNTP, or other forums would only serve to
Kevin Walsh wrote:
The perpetual agreement grants the owner a non-cancellable right
to use changes and/or enhancements made to the Asterisk codebase as
[the] owner sees fit. As any Asterisk fork would, of course, be based
upon existing Asterisk code, the owner would have the automatic right
to
this is an italian code and works... try it.
[channels]
; -- canale 4 --
language=en
faxdetect=both
musiconhold=default
group=2
canpark=yes
context=inbound
signalling=fxs_ks
usecallerid=no
; echo cancel
echocancel=128 ; range from 32 to 256(=echo 100%)
echocancelwhenbridged=yes ; yes = 400 msec
I'm using a Dell GX270 with a single TE110P, no problems here. Of course I had
to take off the pci aluminum card holder thingy to fit in the half height case,
but it works great.
Daniel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thibault Lamy
What do you recommend for E1 and Analog (ala TDM400p)?
Have you tested Dell with other cards yourself?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Andreas Sikkema
|Sent: Viernes, 22 de Julio de 2005 09:37 a.m.
|To: Asterisk Users Mailing List
We are using a Dell PE SC1420 as asterisk server with
one Beronet QuadBRI card (bristuff) and one Digium TE110p and it
works well. No IRQ conflict.
Thib.
Elio Rojano wrote:
DELL computers ussualy has got IRQ conflicts with the USB and slots PCI.
If you disable the USB controller from BIOS
Aidan Van Dyk wrote:
So what are they planning on doing with the Google Summer of Code results?
http://code.google.com/summfaq.html#what_licenses_will_i_have
What licenses will I have to choose from?
This depends on your mentoring organization. For instance if Google
is your mentoring
Juraj Bednar wrote:
Hello,
I just got my Soekris 4801 box for use with Asterisk, but not as a
primary Asterisk server.
* [EMAIL PROTECTED] (Is @home or regular better?)
If you want to run from CF, I recommend running some distribution
(that does not take much space) and your own
I know. but u can't disable the USB controller always. If u have an
server w/ others functions...
Bruno De Luca Graziosi
DELL computers ussualy has got IRQ conflicts with the USB and slots PCI.
If you disable the USB controller from BIOS you get a perfect server.
I have tried several
Hi, I have also failed the same point. Mine is 5.4-Stable Jul 16, did make
world from 5.3 which works * 1.0.6(?) ports and I did cvsup ports-supfile
again several minutes ago. NG.
--
Zen
Darren Wiebe wrote
Did you do a make clean? I just, as in 1 hour ago, successfully
installed 1.0.9
I need to place a SIP FXO gateway in Central America. I've been looking
at Quintum products, but the prices are about $150/FXO port.
I have a Dell SC400 on the shelf, and I'm considering just installing
Asterisk and two TDM04B cards and shipping it down. Does anyone have
multiple TDM cards
Where is the source code?
It looks interesting, But I´m interested in modifying the app, so, only
with a administration password o even a file, you can configure the app...
Users are quite curious about menus and always are looking to improve
de functionality of the configured software...
Joerg,
You will need to have the agent kicked from the queue for it to properly
pass to the next agent.. In agents.conf under [agents], set
autologoff=10. This will kick them out of the queue(logs them out)
for not answering after 10 seconds.
--johann
Joerg Wolf wrote:
Hi,
I've
Aidan Van Dyk wrote:
Kevin P. Fleming wrote:
The first two statements are true; the third is not.
While you can certainly distribute the code you contribute to us via any
other means you wish (under any other license you wish, including the
GPL), the Digium Asterisk source tree cannot
Kevin P. Fleming wrote:
Kevin Walsh wrote:
The perpetual agreement grants the owner a non-cancellable right
to use changes and/or enhancements made to the Asterisk codebase as
[the] owner sees fit. As any Asterisk fork would, of course, be based
upon existing Asterisk code, the owner would
hi,
what does this mean?:
www*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
0 active channels
1 active call
after some searchs got this:
www*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg
172.22.22.27
what is the best web based managment aplication for asterisk ???
Dante
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i'l lhave the source code posted to the site tomorrow..all day
yesterday and today, we've been making major changes from the feedback
from others...
The asterisk manager interface is killing us! every version of
asterisk outputs different events and information..so we're trying to
see all the
I have a VoIP provider that terminates to a PSTN 10-digit number. I
want to be able to send/receive SMS messages that appear to be from
the 10-digit number that my VoIP provider gives me.
I am currently integrating Asterisk with Kannel and it works great.
Unfortunately, my mobile phone
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