[EMAIL PROTECTED] writes:
> 7) Provide the resulting sound files as a free download from your
> website so that others don't have to do the same thing.
In fact, a library of multiple language versions of the standard texts
would be a cool thing for us to build. And, suddenly, the phrase "My
hov
I am developing a sip user agent i am having a problem
with my Callee..When i call from SJphone to my user
agent with Asterisk as the Sip Proxy, it does not
recognize by Ringing and Call answer messages.
___
Do you Yahoo!?
Declare Yourself - Regis
Seems to be alot of these questions on the mailing list recently. AUSTEL
is the old name for the ACA, A-tick is the correct term for certification.
It's only illegal if you connect to a carrier network without A-tick
(you can get consent from them to connect without A-tick).
The ACA has plently
On Thu, 16 Sep 2004 16:12:10 -0700 (PDT),
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Could anyone with any experience with * over a non-PRI T1 help this newbie?
> I have a fractional T1 that is working fine through a channel bank, but I
> can't get any response on * using a T400P. My analog lin
Thanks Paul. I've been getting conflicting information about Austel
permits.. Can any one confirm that the card connecting Asterix to an
existing PABX does not require Austel approval?
Therefore, I could use a simple 1 port Compatible X100P FXO Card (that
doesn't have Austel approval)?
Thanks.
The TDM400 is used for both PSTN and PABX -> PABX connections, from memory.
The card only requires an Austel permit if it is to be connected to an
outside line, from memory.
Cost wise, you can get the TDM400 with 1 line for less than $200, or about
$500 with 4 lines hooked up to it.
Later,
Pau
No, this won't work with Asterisk because of the lack of subscribe /
notify capability.
You might be able to get it figured out with ser.
John
Adam Goryachev wrote:
Just wondering if anyone has gotten instant messaging working between
two polycom phones/and or MSN messenger with asterisk in the m
Hi Paul,
I have yet to find out the make and model of the PABX, I was just doing
some general background research at this point. Is the same device
(TDM400?) used for connecting both to PSTN and to other PABX's? I don't
need to connect the Asterisk box to PSTN at this point, just to another
PABX.
Good to see another Australian user on the list!
You could set up a card with some FXO ports (TDM400?) and use those lines to
hook up the Asterisk box to your existing PABX. But I am sure someone else
will come up with a _much_ more clever solution.
Later,
PaulH
Melbourne
-Original Messag
Currently I have a predictive dialer that is web-enabled, as well as a
superdialer mechanism. I was wondering what kind of success people have
had with their dialers (lessons learned, etc.) and how I need some
direction on how to seamlessly integrate an inbound system with the
predictive dialer.
M
On 16 Sep 2004 at 21:31, Mark Phillips wrote:
> Looks like I've drawn the short straw here.
>
> I do have the facilities and so can do a Male Southern England
> recording but I'm still stuck for female (which seems to be customers
> preference). I also have the techincal know how as well as a web
Title: Creating conference calls from within Astman.
Dear All,
I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call.
I know that within Astman I can define an extension and then origina
Greetings -
We've a pair of ISDN BRI that we use for dialtone, fairly happily except
for the recent meltdown of one of our Netgear RT338's. We're in the
middle of slowly migrating to a VoIP/Asterisk-on-FreeBSD based phone
system. I had originally considered just buying a Digium TDM400 card
and c
What type of existing PABX do you have (Make and Model)
What interfaces can you use to connect to your PABX, ie
analog tie lines, E1/ISDN, anything else?
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of P J
Sent: Friday, 17 September 20
Hi,
I'm new to Asterisk, and am researching information on linking Asterisk
to an existing PBX. Could somebody please help me with what might be
required for the following setup? -
- We have an existing PBX.
- I am going to setup Asterisk on our internal network along with some
internal SIP phone
Thanks for the info, however, I have no idea how to configure
extenions.conf. For example, for 1 block of 10 DID.
1) I want to redirect the call from 1 particular DID number to a particular
FXS port or to a SIP Phone.How can I do that? If I am not wrong, the
incoming call to a particular DID nu
Christopher L. Wade wrote:
The subject says it all.
Is it possible, code wise, configuration wise, at all - to reverse the
order in which the available zap channels are used for *outgoing* calls?
It would be great if there was a config option to select channels via
multiple ways (ie, Round Robin,
I'm guessing that I need more info entered into the 'message centre'
section.
What did you key in?
Paul Hales
IT Support
Adairs
-Original Message-
From: Jeff Pyle [mailto:[EMAIL PROTECTED]
Sent: Friday, 17 September 2004 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discus
The subject says it all.
Is it possible, code wise, configuration wise, at all - to reverse the
order in which the available zap channels are used for *outgoing* calls?
Code wise, I looked at the channel structure and it appears as though
there is only a next pointer, not a previous pointer, so to
Sorry if the last message was sent multiple times. Hit wrong keyboard
sequence in my mailer.
Chris
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Looks like I've drawn the short straw here.
I do have the facilities and so can do a Male Southern England recording
but I'm still stuck for female (which seems to be customers preference). I
also have the techincal know how as well as a web server.
OK folks, I'll start with the common things lik
I have two IP 500's on my Asterisk PBX. The IM features just kinda
worked, without any extra configuration. Messenging and presense.
I've heard of folks trying to interface this functionality with MSN
Messenger and such, without much success. Can't help you much there.
- Jeff
On Fri, 17 Sep
The Polycom phones have an instant messaging function - any idea what is
required to make it work?
PaulH
Adairs
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On 16 Sep 2004 at 20:37, Mark Phillips wrote:
> No disrespect to Alison (whom I know is a Canadian) intended but her
> "British" accent is exactly that; "British". It's very easy to hear
> that she's not from Chipping Sodbury.
>
> Also, do you really have the budget to spend on having all the
> r
I thought about the TTS route. MS have a fairly good set that allows other
peoples engines to be added. The AT&T plugin is quite good.
Perhaps I'll start there and post a few for you all to try.
Still no Taff speakers :-{
Bill Seddon said:
> I agree! Rhetorical (www.rhetorical.com) have a real
Quick question for the experts: I'm seeing stuck SIP channel(s)
scheduled for destruction stay open. This appears to happen
after (apparently successfully) registering with a SIP peer.
Any ideas where to start digging into this? I'm running today's
CVS, however the problem existed before and does
Apple Quicktime will play gsm files iirc.
Rodolfo Grave wrote:
You can use WinAmp or xmms... it has a Plugin for playing GSM files.(not
included in the standard installation but you can find it in google)
RODOLFO
Sys.Concept wrote:
How to play GSM files?
I want to go through some of them but I'm
No disrespect to Alison (whom I know is a Canadian) intended but her
"British" accent is exactly that; "British". It's very easy to hear that
she's not from Chipping Sodbury.
Also, do you really have the budget to spend on having all the relevant
files recorded at $12 a time. That works out to a l
On 15 Sep 2004 at 15:10, Angel Diaz wrote:
> Hi all,
> I have problems with rxfax application. It seems to be ok but I
> don't receive the fax in my directory.
> My extension.conf is as follow:
>
-- SNIP --
> And my log is :
> -- Executing Answer("Zap/1-1", "") in new stack
> -- Executing
I agree! Rhetorical (www.rhetorical.com) have a really good Text-to-speech
system (good in the sense that its voice rendition is quite good). Much
better than Festival or Cephstral (IMHO). The advantage of a good TTS is
that it is possible to have control over exactly what's said, it can be
chan
HI,
I am new to Asterisk and I am setting it up on a RH 9 box. I plan on using
the SIP portion and I have already signed up with FWD for an account. My
question is I have never setup an Avaya 4624 ip phone configuration and I
wanted to take advantage of all the feature that I can by using the co
Talk to Alison Smith she is not an American - She is Canadian!
Then ask her to be whatever you want her to be. Even a Brit.
http://www.theivrvoice.com
> Am I just ranting here or does someone get my point?
>
> --
> Mark Phillips, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com/
> ___
1. Should the r option of the Dial command always generate
a ringing until the called party answers. I have such a scenario but the r
option is not generating a ringing, when I use the m option however I do hear
music. This does not seem correct.
2. Having read the docs etc is it correct th
Could anyone with any experience with * over a non-PRI T1 help this newbie?
I have a fractional T1 that is working fine through a channel bank, but I
can't get any response on * using a T400P. My analog line and the extensions
work fine, but the T400P port 1 shows a red led (what exactly does that
Hi folks,
Does anyone have any "English" voice files rather than "American" voice
files. I know that Digium and Alison Smith have worked hard to provide a
library of sounds etc but this doesn't work for my UK client.
Ideally I'm looking for female files but I'll settle for male ones.
If not the
Hi Folks,
Anyone know how to make a grandstream phone work against a * server when
it is behind a cheap linksys type firewall? I have no control over the
firewall but am allowed to go anywhere I want.
On the * end of the link there is another linksys type firewall which I do
control. What I don't
On Thu, 2004-09-16 at 15:57, Steven Critchfield wrote:
> On Thu, 2004-09-16 at 16:48, Sys.Concept wrote:
> > How to play GSM files?
> > I want to go through some of them but I'm not sure which player to use.
>
> use sox to put a wav header on them.
>
> sox file.gsm file.wav
> sox shouldn't recomp
[EMAIL PROTECTED] a écrit :
attempting to get asterisk pbx to receive inbound faxes
have defined the necessary extension as per technote;
default]
; Answer the line and listen
exten => s,1,Answer
; Dial an extension, let asterisk give a ringtone
exten => s,2,Dial(IAX2/3987,40,r)
; Hangup if nob
I'm not sure you can. Isn't the problem to do with the slicing of the
data. Ulaw does it such that a fax can survive but others don't?
If someone else knows better then I'd love to know how. I want to change
my upstream codec to GalaxyVoice which is currently ULAW for something
skinnier like G729.
On Thu, 2004-09-16 at 16:48, Sys.Concept wrote:
> How to play GSM files?
> I want to go through some of them but I'm not sure which player to use.
use sox to put a wav header on them.
sox file.gsm file.wav
sox shouldn't recompress. It will create add somthing like 40 bytes to
the begining of the
You can use WinAmp or xmms... it has a Plugin for playing GSM files.(not
included in the standard installation but you can find it in google)
RODOLFO
Sys.Concept wrote:
How to play GSM files?
I want to go through some of them but I'm not sure which player to use.
---
avast! Antivirus: Outboun
How to play GSM files?
I want to go through some of them but I'm not sure which player to use.
--
#Joseph
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I have seen this. In order to get PrivacyManager to work with
BroadVoice, I had to write a macro (below) that erases in incoming CID
number if what's pushed from BroadVoice starts with 147 or 192. I
believe the whole issue in general has something to do with BroadVoice
not setting the privacy bit
Let's say you were wanted to terminate
calls onto your Asterisk system but your only available codec was G.729
and you had no control over the remote SIP proxy sending you the traffic.
What would you do?
Does anyone have an update on Asterisk
supporting T.38 with SIP?
Thanks!
chris
Create a new message when starting a new thread. This has nothing to do
with the Earthlink SIP P2P file sharing you responded to.
On Thu, 2004-09-16 at 14:09, Bartosz Wegrzyn wrote:
> Hi,
>
> How can I create a submenus in extensions.conf.
>
> For example:
>
> 1 for english, 2 for polish
>
>
Danny Zak wrote:
Hello Asterisk list;
when i DIAL(H323/[EMAIL PROTECTED]) i get this strange error
--
-- Executing Dial("SIP/home-0953", "H323/[EMAIL PROTECTED]|5|r") new stack
backupns*CLI>
Disconnected from Asterisk server
--
Asterisk just goes down..
You are going to have to provide more debug
Hello Asterisk list;
when i DIAL(H323/[EMAIL PROTECTED]) i get this strange error
--
-- Executing Dial("SIP/home-0953", "H323/[EMAIL PROTECTED]|5|r") new stack
backupns*CLI>
Disconnected from Asterisk server
--
Asterisk just goes down..
--
Best regards,
Danny mail
All good information, thanks. However this is private network between
Asterisk and a Norstar MICS about six feet away. So I'm holding both ends of
the link.
:-)
> -Original Message-
> From: David Troy [mailto:[EMAIL PROTECTED]
> Sent: September 16, 2004 4:57 AM
> To: Asterisk Users Mailin
Excellent! Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Workman
Sent: Wednesday, September 15, 2004 10:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] ztdummy on Fedora Core 2
Ya I ran i
make sure to use the newest firmware.
klaus
el Flynn wrote:
Hi all,
I'm planning to purchase the SPA-2000 to hook up two of our fax machines
to *. Has anyone had any problems using the Sipura for this purpose? Any
"gotchas" i might need to be aware of?
Thanks in advance.
Flynn
__
I sent a test fax last night from home to my office. I sent it through an
ata186 out Asterisk to NUFONE on an iax2 connection. It reported failed. I
resent it with success. When I arrived at my office the fax had been
received both times. I was not expecting this so I don't have any debug or
log
E1 = 32* 64 channels = 2.048MB
Actually gives 30 channels used for voice and one -
frame alignment, the last signalling. In data
applications its 32 clear 64kb wide channels, all
data. (32*64=2048)k
Arinze
--- Bruce Komito <[EMAIL PROTECTED]> wrote:
> You can't run E1 on a circuit designed fo
Nothing special about my config, I am not doing any fax detection
just have a DID
Set up with a triple ring that my fax unit is set to pick up on.
Get this some of my outbound faxes seem to go thru even though it
is reporting an error.
I think its messing up on disconnect.
Paul Seniuk
Does anyone have trouble with dialing in to an Asterisk Server and
having the DTMF digits recognized? We have some clients who are calling
in with cell phones, notably those with SprintPCS service, who's DTMF is
just never recognized.
I have tried relax_dtmf on and off, with no improvement. M
On Thu, 16 Sep 2004, Marcelo Pacheco wrote:
> I'm seriously thinking about developing a trunking VPN utility that would alow
> me to add trunking outside asterisk's code, so I can keep jitter buffer.
We'll fix trunking+jitter-buffer post v1.0
Steve
___
Hi,
I think you need this:
[mainmenu]
exten => s,1,answer
exten => s,2,playback(english_polish)
exten => 1,1,Goto(english,s,1)
exten => 2,1,Goto(polish,s,1)
[english]
exten => s,1,playback(english)
exten => 1,1,goto(support_english,s,1)
[support_english]
exten => s,1,playback(support)
You could have it go to a seperate context, or setup different extensions.
Here's a good link : http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu
- Brian
On Thursday 16 September 2004 07:09 pm, Bartosz Wegrzyn wrote:
> Hi,
>
> How can I create a submenus in extensions.conf.
>
> For example:
Thanks
> simple:
>
>
> [menu]
> exten=>s,1,playback(select_1_for_polish_and_2_for_english)
> exten=>1,1,goto(polish,main,1)
> exten=>2,1,goto(english,main,1)
>
> [polish]
> exten=>main,1,playback(selectoneortwo)
> exten=>1,1,goto(sales,main,1)
> exten=>2,1,goto(support,main,1)
>
> and so on ...
>
simple:
[menu]
exten=>s,1,playback(select_1_for_polish_and_2_for_english)
exten=>1,1,goto(polish,main,1)
exten=>2,1,goto(english,main,1)
[polish]
exten=>main,1,playback(selectoneortwo)
exten=>1,1,goto(sales,main,1)
exten=>2,1,goto(support,main,1)
and so on ...
On Thu, 16 Sep 2004 14:09:33 -050
Hi,
How can I create a submenus in extensions.conf.
For example:
1 for english, 2 for polish
and then again depending which option was selected:
1 support
2 sales
3 other
0 operator
Thanks
Bart,
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h
We suffer the same from with outbound using a mediatrix sip/fx box
The connected fax machine dials and during handshake drops the call.
The Iax link is set to use ULAW
Im trying to get asterisk to handle inbound natively, i.e asterisk answer
listens and dumps into a file on the linux box, I read v
On Thursday 16 September 2004 14:42, Andreas Anderson wrote:
> This is BAAAD! Now even SIP get's "tainted"...
SIP's already tainted... nasty-ass protocol, that. :-)
-A.
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This is BAAAD! Now even SIP get's "tainted"...
http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95
_
Surf the net and talk on the phone with Xtra JetStream @
http://xtra.co.nz/jetstream
___
Noah Miller wrote:
Hi -
I'm wondering if any has experience with the Uniden UIP-200 phones. The
product info says that the 8 led buttons at the top are all
programmable. Can they be programmed as separate line appearances (ala
Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is th
> You're looking for a feature called "Take Back and Transfer",
> TBT for short.
I thought it was Two B-Channel Transfer?
> It works by the telco always monitoring the trunks for DTMF
> from your end, for example, the TBT code might be *8. You
> would send *8,12125551212 down the line and the telc
Rodolfo Grave wrote:
Hi and thanks. I added the entry in the /etc/hosts file and it is
working now... I also had to add more parameters at the peers
definition: authname, username
Now..
The problem with this solution is that my hostname and my ip changes
everytime I reset my box (at least).
vrushank wrote:
!
p.s. maybe set your time/date correct
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Hi and thanks. I added the entry in the /etc/hosts file and it is
working now... I also had to add more parameters at the peers
definition: authname, username
Now..
The problem with this solution is that my hostname and my ip changes
everytime I reset my box (at least)... how can I solve th
attempting to get asterisk pbx to receive inbound faxes
have defined the necessary extension as per technote;
default]
; Answer the line and listen
exten => s,1,Answer
; Dial an extension, let asterisk give a ringtone
exten => s,2,Dial(IAX2/3987,40,r)
; Hangup if nobody picked up within 40
Hello all
We have tested for a mounth or two an asterisk PBX using one T1 channel
bank with 24 fxs and one TDM400P digium card with 4 FXO modules.
This worked with minor problems, the most notorious being some sporadic
static noice or failure in the first FXO module on the wildcard.
Now we have a
On Thu, 16 Sep 2004, Maros RAJNOCH wrote:
> in my * I have one ISDN BRI line with DID (DDI) preselection.
> so in fact I have 100 extensions: +421 33 12 34 56 xx
>
> Q: Is in my power -- or in power of * -- to influence which of these
> extensions will occur in my cellular display?
I guess you m
On Thu, 16 Sep 2004 12:41:47 -0400, Christopher Jacob
<[EMAIL PROTECTED]> wrote:
> When using Asterisk with a PRI to the CO is it possible to transfer a call
> back out and release. In other words, once the call is connected (caller and
> external 3rd party) Asterisk is removed from the equation t
Would anybody have any numbers on how large a box would be required to convert
100 or 200 SIP calls to IAX2, without transcoding, echo cancel, .. Or a setup
with individual IAX2 calls coming on one side, and trunking being used to 1
or more remote boxes on the other side, to improve bandwidth us
If it's what Andrew is talking about, then add the hostname to /etc/hosts.
On Thursday 16 September 2004 05:27 pm, Andrew Thompson wrote:
> Rodolfo Grave wrote:
> > Hi.
> > I cant make SIP calls from asterisk.
> >
> > When I start asterisk, I get the following message: What does it means??
> > Ast
Rodolfo Grave wrote:
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is not behind NAT or Firewall.
--
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.con
Well, you might be better off at that scale to use a cisco as5850 or equiv
with SER and Asterisk. I might not work so well with 672 calls going thru 1
asterisk box.
ds3 <-> Cisco as5850 <-> Asterisk (Possible multiple depending on actual
config and use)
- Original Message -
From: "Marce
Bob Knight wrote:
Steve Underwood wrote:
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
No, but if you find an E3 PCI card with nice Linux support there
might be people interested in helping to get it working with *.
SBE (side band engineering)
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is not behind NAT or Firewall.
--
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
Sep 16 09:5
Steve Underwood wrote:
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
No, but if you find an E3 PCI card with nice Linux support there might
be people interested in helping to get it working with *.
SBE (side band engineering).
--
Bob Knight
[-w
D,
I have a IAX2 gateway that connects to our remote asterisk gateway
that has a PRI.
Inbound seems to work without a hitch. Make sure your iax.conf allows
ULAW as well,
Since fax cannot be compressed.
Outbound is a different story. My fax seems to ring thru, but it never
seems to establish
A
I'm no E1 expert, but as I understand one channel is wasted with framing, so
it is as 2048000 bps link, where one 64000 bps channel is wasted with
signalling. So there's 31 channels left. If you use E&M, FXS or FXO, you
could get 31 voice channels, with PRI or MFC/R2D you get 30 voice channels.
Has anyone had any success using iax for inbound fax into asterisk.
I tried this but can seem to get asterisk to listen for fax, is it zap
specific ?
d
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Hi Again All,
When using Asterisk with a PRI to the CO is it possible to transfer a call
back out and release. In other words, once the call is connected (caller and
external 3rd party) Asterisk is removed from the equation thereby freeing
the PRI channels.
I ask because my scenario is going to r
Even with the robbed bit thing you get 62666.7 bits/s, since it only
steals the LSB every 6 samples. :-)
Regards,
Steve
Marcelo Pacheco wrote:
A T1 is 24 64000bps channels.
The 56000bps thing is when robbed bit signalling is used, it steals bits from
each voice channel for call signalling, while
Christopher L. Wade wrote:
Hi all,
Is there an equivalent of the ${CONTEXT} variable that represents the
*original* context of the call? i.e. If a call originates in the
'internal' context, no matter where it goes, this alternate version of
${CONTEXT} would never change from saying 'internal'?
On Thursday 16 September 2004 12:17, Andrew Thompson wrote:
> Depending on where you using the circuits, you might try an E1. It uses
> the same total bandwidth as a T1(I think), but splits the channels at
> 56K instead of 64K, yielding more channels. (And now I can't remember
> the number.)
uh, n
A T1 is 24 64000bps channels.
The 56000bps thing is when robbed bit signalling is used, it steals bits from
each voice channel for call signalling, while on the E1 one channel is used
for that. When PRI signalling is used each voice channel is the full 64000
bps thing.
Marcelo Pacheco
Em Qui 1
You can't run E1 on a circuit designed for T1. T1 is 24 x 64k = 1.5mb; E1 is 30 x 64k
= 2mb
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 16 Sep 2004, Andrew Thompson wrote:
> Christopher Jacob wrote:
> > All,
> >
> > This may be a stupid question, but her
Andrew Thompson wrote:
Christopher Jacob wrote:
All,
This may be a stupid question, but here it is...
What interface gives the most density? Do I top out at T1's? For
instance, 4
t1's to the Digium Quad span t1 card. Is there an interface available
for T3
or DS3?
Depending on where you using the
Christopher Jacob wrote:
All,
This may be a stupid question, but here it is...
What interface gives the most density? Do I top out at T1's? For instance, 4
t1's to the Digium Quad span t1 card. Is there an interface available for T3
or DS3?
Depending on where you using the circuits, you might try a
Hi again,
in my * I have one ISDN BRI line with DID (DDI) preselection.
so in fact I have 100 extensions: +421 33 12 34 56 xx
Q: Is in my power -- or in power of * -- to influence which of these
extensions will occur in my cellular display?
THANKS.
pgpfkP2ywWN91.pgp
Description: PGP signature
Has anybody any idea what I can do with asterisk following the
Brazilian law.
I do not have a multimedia license or a telecom license, but
I ace asterisk.
Are there companies who are looking for asterisk expertise
in Rio de Janeiro?
Greeting Han
__
Hi everbody,
I have problem with configuring call parking and forwarding.
firstly my setup:
I have one asterisk with gnu-gatekeeper on the same PC.
As phones I use voip-phones with H323 support.
phones are registered on gatekeeper as terminal and
asterisk as gateway.
I setup features.conf (par
All,
This may be a stupid question, but here it is...
What interface gives the most density? Do I top out at T1's? For instance, 4
t1's to the Digium Quad span t1 card. Is there an interface available for T3
or DS3?
Thanks,
Chris
___
Asterisk-User
On Thu, 16 Sep 2004 11:03:48 -0400, Noah Miller <[EMAIL PROTECTED]> wrote:
> > Does anyone have * running on PPC?
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support
>
> Specifically for OS X. There's a download link. The problem still is
> that no one has written ppc d
vrushank wrote:
i cannot see caller ID of the call originated from outside zap channel.
i hv configured both zapata.conf and extensions.conf.
i m right now in india
i think asterisk only supports Bellcore enable caller ID.
so is it the same bug of BT caller ID problem in UK?
or it is the bug of my
i cannot see caller ID of the call originated from
outside zap channel.
i hv configured both zapata.conf and
extensions.conf.
i m right now in india
i think asterisk only supports Bellcore enable
caller ID.
so is it the same bug of BT caller ID problem
in UK?
or it is the bug of my asterisk
Lolll,
That's a good one :))
U make my day :)
Best regards,
Chris HARIGA
P.S.: I send my ethereal log to Intertex.se and I hope to fix the problem
asap. I will post on the list the "solution".
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
- Original Message -
> Message: 3
> Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT)
> From: David Troy <[EMAIL PROTECTED]>
> Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over
> National or DMS100 PRI to a Norstar MICS?
> snip>
> > I have a PRI link up and running between As
> The other issue is that call waiting does not appear to work. The way I'm
> expecting it to work with Asterisk is to send the second call to me - I'm
> using SetGroup and CheckGroup within Asterisk to limit my calls to two at
a
> time total. However, if I'm on a phone call (incoming or outgoing),
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