Hi, I m trying to install [EMAIL PROTECTED] after installing and logging in as root & password i made network connections using netconfig command there i gave ip address as provided by my network provider it displays the ip address I m SORRY to ask that how can i access the net & GUI if u can u
On Wed, Sep 07, 2005 at 10:38:20PM -0500, [EMAIL PROTECTED] wrote:
> Hello
> I have installed asterisk with a card X100P. receives calls but when doing
> the call to the PSTN. says that there are circuits no available... I have
> given to many returns but profit not to make work it I need that they
Are there any standards for setting up pbx dialplans for businesses/offices?
What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses.
If there is such a standard, * and # are
Hello,
Avaya has a nice feature that allows you to
a) ring both a cellphone and a desktop phone at the same time
b) transfer calls (and access other PBX features) from the cellphone that
recieved the call, as long as the call is bridged through the PBX
c) while talking on the cellphone, pick up
Brian Capouch wrote:
> Olle E. Johansson wrote:
>
>>
>> Try setting _ALERT_INFO
>>
>
The reason for this is that if you set *any* variable with one
underscore prefixing the name, that variable will be copied to the new
channel created by dial() - without the underscore. If you create a
variable c
Hi,
I have 4 analogue
PSTN lines on my legacy PBX, 2 lines on one number in a rollover group ZAP1
& ZAP2 and 2 lines on another number ZAP3 & ZAP4.
Is it possible to
have a group of phones ring when lines ZAP1 & 2 are called and a DIFFERENT
set of extensions ring when ZAP3 or ZAP4 receiv
Is there a way of increasing the delay before asterisk
picks up the incoming PSTN call?
I'm using a tdm400p with fxo card. It seems to pick up
the inbound call immediately. I want to delay
detecting the call by about 10 secs if poss.
Done some searching but couldn't find anything
relevant.
Cheer
Thanks Tzafrir and canuck15 for your comments.
Yes I don't think the NIC will be saturated, and I'll search the quality of
the Onboard RAID. I guess I have to learn more about canuck15's comments
though, because I am actually questioning what happens to the board when
you're adding onboard per
taf taffey wrote:
> Is there a way of increasing the delay before asterisk
> picks up the incoming PSTN call?
>
> I'm using a tdm400p with fxo card. It seems to pick up
> the inbound call immediately. I want to delay
> detecting the call by about 10 secs if poss.
>
> Done some searching but could
On Thu, 2005-09-08 at 12:01 +0300, Soner Tari wrote:
> Thanks Tzafrir and canuck15 for your comments.
>
> Yes I don't think the NIC will be saturated, and I'll search the quality of
> the Onboard RAID. I guess I have to learn more about canuck15's comments
> though, because I am actually questio
asterisk-users
who use astlinux with booting from DOM?
how to do ?thanks
oncemore
[EMAIL PROTECTED]
2005-09-08
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Hello,
I think I might have an
inkling as to where the issue may be at. For some reason when I create a
new context, a directory is not created in /var/spool/asterisk/voicemail.
The default context resides there but new ones are not created.
Has anyone ever experienced this or does
any
If I use the wait command won't this intercept the
inbound call?
I want the call to stay on the pstn line for 10
seconds before asterisk detects the inbound call.
Taff.
--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
> taf taffey wrote:
> > Is there a way of increasing the delay before
> ast
Sorry about my previous post, "outbound routing" this was clearly
available in AMP.
However,
How would I go about making all calls from PSTN line 1 (X100P #1) ring
call group #1, PSTN line 2 (X100P #2) call group #2, and the SIP
incoming calls route to group #1 also. Is this done via dial plans?
hello,
i installed an asterisk as a pri gateway. Everything is okay.
/etc/init.d/zaptel starts successfull with wct4xxp module.
/etc/init.d/asterisk starts also successfully. I tested my pri cable and
it works. But still my span isn't up. I don't see any error. Do you have
any idea? What els
Hi,
I would like to use the * when I am in the asnwer machine, but I received a message asking for the temporary pass code.
Where I need to put this pass?
I am using asterisk 1.2.0 beta 1
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On Thursday 08 September 2005 05:38, taf taffey wrote:
> If I use the wait command won't this intercept the
> inbound call?
Why not try it and see?
> I want the call to stay on the pstn line for 10
> seconds before asterisk detects the inbound call.
You can't prevent Asterisk from detecting the
Will try it later when i get a chance..
Cheers for the input..
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
> On Thursday 08 September 2005 05:38, taf taffey
> wrote:
> > If I use the wait command won't this intercept the
> > inbound call?
>
> Why not try it and see?
>
> > I want the call
hi all,
i've this problem with app_txfax. Here's the log of the error:
Sep 8 13:28:55 VERBOSE[10750]: -- Attempting call on Zap/g1/2430 for
application txfax(/var/tmp/ast_fax-1126178934.10240.1804289383.0|caller)
(Retry 1)
Sep 8 13:28:55 DEBUG[10750]: Using channel 3
Sep 8 13:28:55 WARNIN
what about a copy of your zapata.conf and zaptel.conf,what color is the
leds
On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote:
> hello,
>
> i installed an asterisk as a pri gateway. Everything is okay.
> /etc/init.d/zaptel starts successfull with wct4xxp module.
> /etc/init.d/asterisk sta
Hello,
I'm still looking for any ideas on this problem:
I've got 3 sip clients behind the router, and they all register with
Asterisk using the same IP address.
Now, wenn all are registered, all the calls get routed to the client that
registered most recently, but not to the correct client.
Als
hi,
my asterisk version is 1.0.9
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
it is comfortable with Turkish Telecom. i tried before and it works.
/etc/asterisk/zapata.conf
[channels]
switchtype=euroisdn
signalling=pri_cpe
context=incoming
group=1
channel=>1-15,17-
Just one line. Do you think you could point me to the SPA3K. A google
search doesn't yield any results. Is that a discontinued product? Would
I not need something on the other end where the POTS phone line is
located? Thanks!
--
Patrick Campbell
Hi
i know that we can use sip clients through nat, like the same way can we use
sip clients through a proxy,.
is there any sip client that i can specify a proxy address and use or any
sip device.
regards
Kanishka
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Hello everybody
This question has probably already been asked, but I'd like to have
feedbacks about Yuxin hardphones
Especially series 10, 100 and 200 ( and by the way i didnt found too
much technical differences between those models). Is it better than
budgettone, or so cheap hardphones ?
I m
> Given the current discusison regarding ztmonitor, line testing, etc., I've
> been looking into purchasing a used transmission test set. From my
> research, it seems that there are two items that might fit the bill: the
> HP 3551A and the HP 4935A.
>
> I know nothing about these specific de
Would like to be able to do the following, which is typical of an
internal setup:
On handset pickup get some kind of internal
dial tone (probably user defined in indications.conf) Pressing 9, for dialing an external line, to immediately
switch to the PSTN dial tone. continue with
dialing t
> Just one line. Do you think you could point me to the SPA3K. A google
> search doesn't yield any results. Is that a discontinued product? Would
> I not need something on the other end where the POTS phone line is
> located? Thanks!
spa3k is really an spa3000 (k = 000).
Try:
www.sipura
On Thu, 2005-09-08 at 13:54 +0300, Dias Badekas wrote:
>
> Would like to be able to do the following, which is typical of an
> internal setup:
>
> 1. On handset pickup get some kind of internal dial tone
> (probably user defined in indications.conf)
> 2. Pressing 9, for dialing
i have a box running debian sarge with asterisk installed from distribution
packages:
CLI> show version
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64
running Linux
I have managed to configure a simple dialplan (the PBX task is quite simple as
this is a small offic
I've been messing with it for a couple weeks with MySQL. It seems pretty good to me though I have had a couple crashes. I cane' say for sure that the crashes were directly related to RealTime though. Also, I'm still using CVS HEAD 2005-09-06 which was right before the beta release, I think.
O
Not that it's very widely used, but I thought it worth mentioning, if
you intend to use TDMoE with multiple Asterisk servers locally your
ethernet will be fairly well saturated and you will want a second NIC
connected to a separate isolated network for your TDMoE trunks.
MATT---On 9/8/05, Dave Cot
On Thu, Sep 08, 2005 at 02:56:28PM +0200, Marek Zachara wrote:
> i have a box running debian sarge with asterisk installed from distribution
> packages:
>
> CLI> show version
> Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64
> running Linux
>
> I have managed to conf
Flobi wrote:
I've been messing with it for a couple weeks with MySQL. It seems
pretty good to me though I have had a couple crashes. I cane' say for
sure that the crashes were directly related to RealTime though. Also,
I'm still using CVS HEAD 2005-09-06 which was right before the beta
rele
I've being using the same non-nat config for a 7960 for about a year with no
issues.
I have just upgraded cvs head from the 16th of July to todays.
About a minute after I make a call I get the message
chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission
[EMAIL PROTECTED] for seq
If you're talking about asterisk actually doing anything with the call on a logical basis (i.e. processing your dialplan), "wait" will halt that. Actual detection (as recorded in CDR) and acknowledment (via the lower level SIP/IAX/etc to the requester) begins when the call is received and the "wai
b is possible. See res_features.conf for more information on transferring via DTMF.
c is not yet possible. This would require shared call appearances which isn't yet implemented.
On 9/8/05, Arnar Birgisson <[EMAIL PROTECTED]> wrote:
Hello,Avaya has a nice feature that allows you toa) ring bot
Generally I have used Intel Chipsets on ASUS motherboards. I've always
used Kingston RAM. I've used Intel P4 CPU on S478 and LGA775.
The Asus boards almost always have NIC and sometimes on board VGA. I've not
had any problems with the hardware.
Regards,
Chris
- Original Me
Hi all
I am looking at implementing asterisk at a company with two ISDN bricks (60
lines). I know that the VoIP will absorb at least on brick worth of lines but
that still leaves me with a need for 30 ISDN lines. As far as I can tell most
of the Digicom cards have 4 FXS ports and I've read on t
Hi,
is there anyone trying a power over ethernet solution to feed IP phones?
I'd like to buy a "good but cheap" hub/switch but I don't know which.
Can anybody help me??
TIA
Giorgio
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I'm trying to setup my Asterisk-PBX (using [EMAIL PROTECTED]) to work as
an ISDN-VoiP-Gateway. For this to work, I'm trying to setup a trunk
using my Billion PCI-ISDN-Card. And yes: I'm an Asterisk-Newbie. ;-)
I can't get it working and was wondering if there is someone out there
with the same set
Are you looking for a Mid-span hub or a switch? Basically a Mid-span
hub will function as a Powered "patch panel" where a switch will be just
that. A switch that will feed power to each device.
What do you consider cheap and how many ports are you looking for?
Regard
Hi
We are
using an IVR system based in AGI.
The
AGI makes querys to our radiator server using radius
libraries.
It is based on ASTCC and Net-Radius
modules.
Look
this links:
http://www.voip-info.org/tiki-index.php?page=ASTCC
http://search.cpan.org/~luismunoz/Net-Radius-1.44/
Regards
hi all i have a problem, with digit 9. to go outside asterisk give me the error "All Circuits are busy" what happend??
my extension.conf
[globals]
VM_PREFIX = *
RINGTIMER = 15
REGTIME = 7:55-17:05
REGDAYS = mon-fri
RECORDEXTEN = ""
PARKNOTIFY = SIP/200
OUT_1 = ZAP/g0
OPERATOR =
NULL = ""
IN_OVERR
On Thursday 08 September 2005 08:33, Chris wrote:
> Generally I have used Intel Chipsets on ASUS motherboards. I've
> always used Kingston RAM. I've used Intel P4 CPU on S478 and LGA775.
> The Asus boards almost always have NIC and sometimes on board VGA. I've
> not had any problems
that still leaves me with a need for 30 ISDN lines. As far as I can tell most
of the Digicom cards have 4 FXS ports and I've read on this list that at most
two could coincide in a box simultaneously without causing an interupt flood.
Is it true ?
My boss is just asking me if it is possi
Sorry about my previous post, "outbound routing" this was clearly
available in AMP.
However,
How would I go about making all calls from PSTN line 1 (X100P #1) ring
call group #1, PSTN line 2 (X100P #2) call group #2, and the SIP
incoming calls route to group #1 also. Is this done via dial plans?
On Thu, 2005-09-08 at 11:07 -0300, Pietro U wrote:
> hi all i have a problem, with digit 9. to go outside asterisk give me
> the error "All Circuits are busy" what happend??
>
> [outrt-002-9outside]
> include => outrt-002-9outside-custom
> exten => _1XX,1,Macro(dialout-trunk,1,${EXTEN},)
> exten
On Thursday 08 September 2005 10:26, Simone Cittadini wrote:
> Is it true ?
> My boss is just asking me if it is possible to stuck 4* TE411P in a
> single server, for a total of 480 lines, someone can assure me it is
> possible/impossible (manageable/unmanageable) from real-life experience ?
Don't
I know I have seen something on the mailing list describing how to run
more than one instance of Asterisk. I can't find it anymore.
What are the things to look for when running more than one copy.
Yes, I know about contexts.
thanks,
Geoff
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gincantalupo wrote:
> Hi,
> is there anyone trying a power over ethernet solution to feed IP
> phones? I'd like to buy a "good but cheap" hub/switch but I don't
> know which. Can anybody help me??
>
We are testing out the 3Com 2226-PWR Plus ($800US roughly). We haven't made
it too far but the pho
We use the old Cisco 3500, and works fine. The issue is which phones are you
going to use, in order to have no problems with the polarity and the wiring.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gincantalupo
Sent: Thursday, September
On Thursday 08 September 2005 10:26, Simone Cittadini wrote:
>> Is it true ?
>> My boss is just asking me if it is possible to stuck 4* TE411P in a
>> single server, for a total of 480 lines, someone can assure me it is
>> possible/impossible (manageable/unmanageable) from real-life experience ?
This is an old solution for pbx's, was called autopatch.
The issue is that the pbx is full duplex, and most of this radios are half
duplex.
That is all, so the rpt solution deals with that plus the electrical
interface to the system.
Carlos Alperin
-Original Message-
From: [EMAIL PROT
This is really not a huge help.. hut I got that exact error message
running on a pentium 1.8GIg, 128meg of ram, CentOS 3.0, self compiled
CVS-HEAD of 2.0.
The crash message is less them useful. When it happens you can not
do anything with asterisk and have to kill it.
On 9/7/05, stevanus <[EMAI
On Thursday 08 September 2005 16:26, Simone Cittadini wrote:
> My boss is just asking me if it is possible to stuck 4* TE411P in a
Doesn't that equal 16 lines, not 480 lines? Or did I miss something?
--
Regards
Wayne Gemmell
Tel & Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECT
Hi,
I think about 8 ports. It is for big firms so I think about 20-25$ for
each port (to tell the truth I don't know how much a POE hub can cost,
cannot find any price list on internet...).
TIA
Giorgio
Sean Milheim wrote:
Are you looking for a Mid-span hub or a switch? Basically a Mid-spa
Dave thanks for the reply. in my PBX (digital phones) i dial 9 and get
outside line and local numbers have 7 digits. is the conf correct for
this numeric plan?On 9/8/05, Dave Cotton <[EMAIL PROTECTED]> wrote:
On Thu, 2005-09-08 at 11:07 -0300, Pietro U wrote:> hi all i have a problem, with digit 9.
Geoff Karl wrote:
I know I have seen something on the mailing list describing how to run
more than one instance of Asterisk. I can't find it anymore.
What are the things to look for when running more than one copy.
Yes, I know about contexts.
thanks,
Geoff
This begs a repeated question: W
I am not able to get softphone registered (active) with * .
new installation , new user
Able to get server started , and phone appears to register …
gets the SIP reponse 481 message
Register SIP ‘4009’ at 192.168.200.10 port 2199
expires 120
Unregistered SIP ‘4009’
Regi
We have 14 Asterisk servers with Asus/Intel in production at our four
locations. We very much recommend them and also go through
zipzoomfly.com to buy the parts.
MATT---On 9/8/05, Martin <[EMAIL PROTECTED]> wrote:
On Thursday 08 September 2005 08:33, Chris wrote:> Generally I have used Intel C
We are drafting a plan for a new office setup. The users will be using Cisco
7940 phones registered to a remote Asterisk server. We were thinking of
using two ADSL lines coming into a Multi-WAN router to allow for load
balancing. As opposed to setting up half the users on one ADSL line, half on
the
In article <[EMAIL PROTECTED]>,
Wayne Gemmell <[EMAIL PROTECTED]> wrote:
> On Thursday 08 September 2005 16:26, Simone Cittadini wrote:
> > My boss is just asking me if it is possible to stuck 4* TE411P in a
> Doesn't that equal 16 lines, not 480 lines? Or did I miss something?
16 E1 trunks, each
On Thu, 2005-09-08 at 16:26 +0200, Simone Cittadini wrote:
> Is it true ?
> My boss is just asking me if it is possible to stuck 4* TE411P in a
> single server, for a total of 480 lines, someone can assure me it is
> possible/impossible (manageable/unmanageable) from real-life experience ?
>
Y
Hi,
I found some contradicting infos about pass through of
T.38 data.
Are there any experiences of just passing T.38 via SIP from one T.38
application or gateway trough asterisk to another T.38 application
or gateway?
Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip
(without unde
Maybe this could be used with the Internet Repeater trunking system I
primarily use VHF... But would be interested in setting that up on my asterisk
with the Internet 2M Repeater trunking system
Robert A. Huddleston, KF4BYY
Cavalier Telephone LLC.
(Desk) 804.422.4401
(Cell) 804.400.3686
[EM
Wayne Gemmell wrote:
On Thursday 08 September 2005 16:26, Simone Cittadini wrote:
My boss is just asking me if it is possible to stuck 4* TE411P in a
Doesn't that equal 16 lines, not 480 lines? Or did I miss something?
Yes, you missed something:
4 PRIs = 92 Lines per Card * 4 Card
Wayne Gemmell wrote:
Hi all
I am looking at implementing asterisk at a company with two ISDN bricks (60
lines). I know that the VoIP will absorb at least on brick worth of lines but
that still leaves me with a need for 30 ISDN lines. As far as I can tell most
of the Digicom cards have 4 FXS p
Andrew Kohlsmith wrote:
On Thursday 08 September 2005 10:26, Simone Cittadini wrote:
Is it true ?
My boss is just asking me if it is possible to stuck 4* TE411P in a
single server, for a total of 480 lines, someone can assure me it is
possible/impossible (manageable/unmanageable) from real-life
We run many servers with 4 Quad cards and have no problems, SANGOMA works
great for this !!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
groups
Sent: Thursday, September 08, 2005 6:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discus
Hi,
Does someone know what the problem is when the
call goes through but one or both parties can't
hear the other?
What are the common causes? Solutions?
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Asteris
On Thursday 08 September 2005 13:12, asterisk groups wrote:
> You might want to offload some of that PRI termination to an external
> device such as a Cisco AS53XX, Lucent MAX TNT, Audio Codes or Redfone
> fonebridge device and then trunk it to your Asterisk servers. But
> putting more then 2 quad
On Thu, 2005-09-08 at 11:53 -0300, Pietro U wrote:
> Dave thanks for the reply. in my PBX (digital phones) i dial 9 and get
> outside line and local numbers have 7 digits. is the conf correct for
> this numeric plan?
Still I'd like to see the Macro, does it strip the 9 off the front?
If not you'r
I've had lots of luck with the Intel/Asus and I am the part supplier.
Chris
- Original Message -
From: "Matt Florell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, September 08, 2005 9:58 AM
Subject: Re: [Asterisk-Users] Motherbo
Hello, all I apologize for all these questions. But I
have so many running through my head for this. What
brand of server is a good one to use for running an
asterisk box? I did an install the other day on a
Compaq Proliant ML370 G2 / Debian Sarge and it is
currently working great. But HP/Compaq
Is there some way to know if the fax was received correctly or not?
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To UNSUBSC
I apologize for the double post. I am curious as to
what the usefullness is of the multiple line
appearance feature on Polycom phones. I setup our
phones to register one line per extension but I hear
the IP501's can do three line appearances. Why and
how could this feature be applied?
Thanks
In article <[EMAIL PROTECTED]>,
Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Wayne Gemmell wrote:
> > On Thursday 08 September 2005 16:26, Simone Cittadini wrote:
> >
> >>My boss is just asking me if it is possible to stuck 4* TE411P in a
> >
> > Doesn't that equal 16 lines, not 480 lines? Or did I
On Thursday 08 September 2005 11:19, Matthew Boehm wrote:
> Yes, you missed something:
> 4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines
> That is assuming you have 1 D-chan per span.
You're also assuming T1.
-A.
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We're now a little more than a month away from AstriCon 2005 - The
Asterisk Conference and Exhibition. We need everyone who plans on
attending to register with the Hyatt ASAP to ensure we have enough
hotel rooms. (Last year in Atlanta we over-booked the hotel by over
Did you use the 1.1.x version of the patch and chan_unicall.c ?
Denis.
On 05 de set de 2005, at 20:57, Anton Krall wrote:
Guys.
Anybody gotten unicall to compile under cvs-head? I get a lot of
errors
while under 1.0.9 everything compiled without a hickup.
Any hints?
___
Michael Welter wrote:
My preferred LD vendor requires g729 and SIP.
Is there a method to test, prior to initiating a call, whether a g729
codec is available? Will ChanIsAvail test g729 availability?
To clarify:
I have g729 licenses for my system. If I have g729 calls in
process then I
I'm not using an FTP server :-/ I guess I'll have to setup an FTP server
and have it get it's files from there...it appears that something is
corrupted on the phone.
...
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday,
I use them with just the one NIC card. I don't use them as a router so
the phones and my gateway are all on the same network.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, September 06, 2005 4:33 PM
To: Asterisk Us
Is it true that asterisk voicemail moves messages from old
to inbox after they age a certain number of days? How many days?
I just had a case where 60 old messages showed back up in someone’s
inbox and the time interval is unknown because I do not know when they were
saved.
__
Ben, That is the correct choice for an Asterisk box. good luck.
Ben Brown wrote:
Thanks for the replys. I'm
convinced. PRI it is.
Peter Svensson wrote:
On Mon, 5 Sep 2005, Ben Brown wrote:
So the only difference with PRI is caller ID? What I am trying to
determ
That's almost always due to a NAT problem. Try using a stun server to solve
this problem (stun.gist.net for example)
->-Original Message-
->From: [EMAIL PROTECTED]
->[mailto:[EMAIL PROTECTED] On Behalf Of Doug
->Sent: Thursday, September 08, 2005 11:38 AM
->To: asterisk-users@lists.digiu
On machine A I have something like the following in extensions.conf:
[iax-extensions]
exten => _9.,1,Dial(IAX2/machineB/${EXTEN:[EMAIL PROTECTED])
exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS})
exten => _9.,3,Hangup
On machineB I have something like this:
[mycontext]
exten => 2002,1,Dial(SIP/2002
ok ok ok. this conf, i copy and paste from internet and i try to
use in my asterisk. what i need to get outside line from the pbx?
in my pbx i dial 9 and the 7 digit numbers.
sorry but im a really newbie.
On 9/8/05, Dave Cotton <[EMAIL PROTECTED]> wrote:
On Thu, 2005-09-08 at 11:53 -0300,
Kenny Kant wrote:
Hello, all I apologize for all these questions. But I
have so many running through my head for this. What
brand of server is a good one to use for running an
asterisk box? I did an install the other day on a
Compaq Proliant ML370 G2 / Debian Sarge and it is
currently working
Jason Becker wrote:
Sage advice, but out of curiousity what happened to Digium's T3 card
(the DS3000P)?
IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC it
will have no on-board EC and no on-board encoding so I can't imagine the
machine you would need to process that many
I was only able to find spandsp-0.0.1k, I am trying to follow the AMP
install guide, it suggests that I use 0.0.2pre18 but the site
(soft-switch) is down. Does anyone have a copy of it or later?
Also when attempting to compile the 0.0.1k I get errors that won't let
me continue, I assume those are
Yes, but on that case, you still are sending digital signal. On the
autopatch you just send analog voice. On the Internet, you 'll going to need
to use IP Phones on the other side, on the autopatch, you will be talking
with someone on the radio (Two way or half duplex, or simplex)
That is the diff
On 9/8/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Geoff Karl wrote:
> > I know I have seen something on the mailing list describing how to run
> > more than one instance of Asterisk. I can't find it anymore.
> >
> > What are the things to look for when running more than one copy.
> >
> > Yes,
Brilliant, thanks for the response.
Arnar
>>> [EMAIL PROTECTED] 8.9.2005 13:26:52 >>>
b is possible. See res_features.conf for more information on transferring
via DTMF.
c is not yet possible. This would require shared call appearances which
isn't yet implemented.
On 9/8/05, Arnar Birgiss
On Thursday 08 September 2005 12:16, Michael Welter wrote:
> I have g729 licenses for my system. If I have g729 calls in
> process then I don't want to attempt another g729 call. Is there a
> method to test whether a g729 codec license is available?
Currently no. This would be a great starter
Matthew Boehm wrote:
Jason Becker wrote:
Sage advice, but out of curiousity what happened to Digium's T3 card
(the DS3000P)?
IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC
it will have no on-board EC and no on-board encoding so I can't imagine
the machine you would
How do I set each extension to play it's own voicemail prompts? I have vm
working in that it plays the standard "person at extension 1234 is not
available." and takes the message. I've recorded seperate .gsm files for
each user but can not figure out how to use them.
- Gary
Edison Informati
On Thu, 8 Sep 2005, Rich Adamson wrote:
spa3k is really an spa3000 (k = 000).
Try:
www.sipura.com
www.voxilla.com
www.voipsupply.com
or any number of other suppliers of voip equipment.
Oh Ok I guess I was taking it too literally!!!
With a pair of SPA3000's, would I not even need *?
--
Pat
can download this http://www.acodin.com.co/util/spandsp-0.0.2pre18.tar.gz
Diego
> I was only able to find spandsp-0.0.1k, I am trying to follow the AMP
> install guide, it suggests that I use 0.0.2pre18 but the site
> (soft-switch)
> is down. Does anyone have a copy of it or later?
>
> Also when
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