On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED]wrote:
[..snip..]
With that many extensions, I'll second using a SIP registrar like
Freeswitch or OpenSer. Just use asterisk to provide the services.
Is Asterisk even needed?
- Gonzalo
On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
[..snip..]
With that many extensions, I'll second using a SIP registrar like
On Fri, Nov 21, 2008 at 2:49 PM, Noah Miller [EMAIL PROTECTED]wrote:
And FreeSWITCH can't handle that?
Freeswitch can provide many PBX features with additional modules, but
asterisk can provide more, and its implementations of such items are
more time tested. One of freeswitch's big
On Thu, Nov 6, 2008 at 2:11 PM, David Gibbons [EMAIL PROTECTED]wrote:
I'm glad I'm not the only one who got that. I sent them a nasty response
earlier this morning...
I got the same crap from them. I can't imagine anyone buying from a company
that spams subscribers of a mailing list to get
On Thu, Aug 7, 2008 at 2:04 PM, Joseph [EMAIL PROTECTED] wrote:
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it is real and not a scam.
The (FWD) does not do anything to
Hi All,
In an effort to avoid storing cleartext passwords for users, I was looking
around to see if iax.conf had a similar setting to sip.conf's 'md5secret'.
It looks like it doesn't. I've set auth=md5 which, by the look of it, makes
Asterisk work out the md5 on the fly of the cleartext password
On Wed, Apr 2, 2008 at 10:28 AM, Greg Woods [EMAIL PROTECTED] wrote:
I've been a happy user of asterisk for over a year just for a small home
setup (a Digium TDM400P with one POTS line and three internal extensions
plus a couple of SIP phones). I recently moved from running Fedora Core
6
Hi All,
For the most part, the PBX works as it should. Occasionally people complain
that they call and the PBX doesn't pick up. Other times it looks like the
call is answered by Asterisk but I still hear ringing and I start listening
to the IVR menu a few seconds into it.
As for Asterisk not
On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
Any suggestions??
I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2.
A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 ?
Yes, I built 1.6.0-beta4 just recently with zaptel 1.4.9.2. As per your
suggestion on IRC,
I think this type of abuse is well deserved due to the way he intended to
advertise his business, so I'll add a bit of wood to the fire. How about
the sign-up form?? Some serious HTML design work going on there.
- Gonzalo
On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote:
The
On Fri, Mar 7, 2008 at 9:52 AM, Faraz Khan [EMAIL PROTECTED] wrote:
It does work. Did you do the switch statement in extensions.conf?
If not check voip-info for Asterisk Realtime Extensions
Hi Faraz,
I just realised I never replied to this message. Yes, you were right. I
simply had to add
Hi All,
This question is probably more for the LDAP experienced users/developers as
I'm sure it would work fine if I weren't using LDAP (but I am, and I'm
almost there with the setup!!!).
I've setup an extension with the following:
AstExtension: 210
AstApplication: Macro
AstApplicationData:
On Sat, Mar 15, 2008 at 8:25 PM, Gonzalo Servat [EMAIL PROTECTED] wrote:
[..snip..]
AstExtension: 210
AstApplication: Macro
AstApplicationData: call-ext,SIP/testuserIAX2/testuser,210
When I dial this extension, I see the following in the log:
-- Executing Macro(SIP/testuser-082b11f8
On Fri, Mar 14, 2008 at 9:01 AM, Rizwan Hisham [EMAIL PROTECTED]
wrote:
I dont know about IAX, but for SIP users you can use the function
SIP_HEADER(headername) to get the information u need from the sip packets.
for example you can use SIP_HEADER(From) which will give you the From header
On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
Do you have canreinvite=no in the sip client configuration? If not then
the two sip phones are probably issuing a reinvite command and taking
asterisk out of the call path. If that happens and the phones can't reach
Hi All,
I'm trying to achieve the following:
- If sip/iax user logs in from home, they can dial internal extensions
only (this is to avoid employees going wild on local/mobile calls from home)
- If sip/iax user logs in from the office, they can call anyone they want.
Since I have my users
Hi All,
I have 2 SIP clients configured and connected to Asterisk. When I place a
call from SIP1 to SIP2, if both codecs are the same then everything works as
expected. I then allowed one of the clients to use alaw instead of ulaw and
there were audio problems (couldn't hear the other end, etc).
On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan [EMAIL PROTECTED] wrote:
Gonzalo,
Please let us know what you mean by 'stops working' - it should spit
out errors or wrong queries to ldap.
Basically what I mean by that is that in the slapd debug, no activity was
going on when I tried to
On Fri, Mar 7, 2008 at 8:46 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan [EMAIL PROTECTED] wrote:
Also please keep this list in your replies. I have no problems
answering personal emails but both of us might get more feedback if we
post our
Hi All,
I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree
where the users will each have their account, SIP username/password,
extension number, context, etc. My first question is: can this be done with
1.4.x? If so, where can I get the res_config_ldap from??
I googled
none at all. Is this
normal?
Thanks!
Regards,
Gonzalo
On Thu, Mar 6, 2008 at 12:37 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
Hi All,
I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree
where the users will each have their account, SIP username/password,
extension
On 10/23/07, Alan Lord [EMAIL PROTECTED] wrote:
Luis Antonio Prata Barbosa wrote:
Hi,
Some days ago I spent about US$700,00 in a Tormenta III board in
www.govarion.com http://www.govarion.com. I used credit card.
I didn't receive any answer for my emails and there is no telephone
On 9/27/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote:
Some company asked me to do audits with there asterisk boxes. Is there a
standard that i should be following in auditing? anyway can give me a
start
what to do with asterisk
On 6/15/07, Kyle Sexton [EMAIL PROTECTED] wrote:
I'm wondering if anyone out there is running a community PBX for their
local Asterisk User Groups or area Linux groups. I've been thinking of
setting one up but am stuck as to what services to provide that people would
actually find useful. I
On 8/13/06, Attilla De Groot [EMAIL PROTECTED] wrote:
[..snip..]
Sorry, didn't thought it was relevant, since the entire macro gets
executed, but here it is.
;recording
exten = _*22*XXX,1,Macro(record,conference,${EXTEN:4))
I think what you probably want is:
exten =
On 8/9/06, Yaakov Menken [EMAIL PROTECTED] wrote:
I really don't understand the complaint. Fonality gets a $5 mil.
investment for building its own system on top of Asterisk -- no
complaint. But Mark Co. can't get VC for their own business /
enterprise / support architecture?
Everything that
On 7/25/06, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
I'm setting up a branch office, but I don't want to trunk from the main
office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the expense
of a second Asterisk
On 7/24/06, Benjamin Stocker [EMAIL PROTECTED] wrote:
Hi!
What's wrong with this?
exten = s,1,Set(myvar=nothing)
exten = s,2,Set(myvar = $[${CALLERID(num)} : ([a-z]+)])
exten = s,3,NoOp(${myvar})
Try removing the spaces on either side of the = symbol.
Regards,
Gonzalo
On 7/20/06, calvis [EMAIL PROTECTED] wrote:
We are looking at various software packages that do Project Management
Collaboration. Since I value the opinions of this list I would be
interested in how others are dealing with Project Management
Collaboration. By collaboration I mean the
On 7/19/06, Matthew Warren [EMAIL PROTECTED] wrote:
We build custom scripts for Asterisk. We can build this for you, for
reletivly inexpensive. But you will need to contact me thru email at
mwarren at procomconsulting dot com .. This is a commercial app you
need but requesting on a non
On 6/11/06, James Harper [EMAIL PROTECTED] wrote:
[.snip.]
My dialplan in the pap2 is:
(:0S0)
Which causes it to dial a '0' to asterisk as soon as I gets picked up.
In my asterisk dialplan it then does a DISA to another context, which
means Asterisk is doing all the dialplan stuff. For what I
On 7/12/06, Jeremy McNamara [EMAIL PROTECTED] wrote:
Michael Workman wrote:
Well you told me to talk to him
And he said he worked for NuFone so its not My Fault Period...
Its your Fault for pushing me to Him and You allowing him to say he works
for NuFone...
If Anything You should Go
On 7/12/06, Alex Robar [EMAIL PROTECTED] wrote:
His work ethic is fine... He either couldn't do the job you wanted, or he
didn't want to, so he sent you to someone else. If you paid Greg instead of
NuFone, then that's really tough shit for you for not following the
instructions Jeremy gave you.
On 6/21/06, Matt [EMAIL PROTECTED] wrote:
Hi,
I'm still in the process of debugging this, but I have a gotoif
statement that looks like this:
exten = 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues,210,1)
exten = 26,n,Goto(ext-local,${VM_PREFIX}127,1)
[..snip..]
Hi Matt,
Are you sure the
Hi Doug,
On 6/9/06, Doug Crompton [EMAIL PROTECTED] wrote:
Replying to myself here... I got the latest 1.2 head via svn. Did a patch
diff'ed it to latest AEL2 (as described at:
http://voip-info.org/wiki/view/Asterisk+AEL2
Patched it. All went fine. On compile I get the following error
On 6/9/06, Joshua Colp [EMAIL PROTECTED] wrote:
[..snip..]
I'd just like to note that AEL2 was brought over into Asterisk trunk
(what will become 1.4) and the old AEL removed. That's where most
development is taking place on AEL2, and why you don't see patches on
the bug tracker.
Hi Joshua,
On 6/4/06, Attilla De Groot [EMAIL PROTECTED] wrote:
Hi all,
I'm trying to make a context that will monitor a call and when it's
completed it would e-mail the wav to a specified mail adres.
So I made a standard context that records a call, like this:
exten =
On 6/4/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming
Hi guys,
I'm interconnecting an Asterisk box with a Lucent Definity PBX by
means of FXO/FXS ports on a TDM2400 card. Everything works well,
except for one little thing. Every now and then somebody (from an
Asterisk extension) will call another extension on the Lucent Definity
PBX and they hit
Hi All,
I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd
PAP2-NA units all hooked up to Asterisk. As you can imagine, setting
them up took a while, and changing settings on them also takes a
while. In order to prepare for future deployments, I'd like to use XML
provisioning
On 4/24/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi Friends,
[..snip..]
--- Employee 1 PC (Softphone i.e., Headphones with Mic)
--- Employee 2 PC (Softphone i.e., Headphones with Mic)
--- Employee 3 PC (Softphone i.e., Headphones with
On 4/23/06, Roshan Sembacuttiaratchy [EMAIL PROTECTED] wrote:
I use the following dialplan within the Sipura:
([2-79]11:@gw0|999:@gw0|112:@gw0|0[12]x.|[*x]xx.:@gw0|#9,:[*x]x.|**)
[..snip..]
Is this @stuff something new in the SPA3000 dialplan syntax? I have
SPA-200x ATAs and I never saw any
On 2/27/06, Darren Ellis [EMAIL PROTECTED] wrote:
Hello,
I have a request from a customer that I'm not sure how to implement.
They have a Snom-360 as receptionist phone and SPA-941 for all other
phones. They use the SPA-941 DND function when they are away from their
desks, which happens
On 2/22/06, Matt [EMAIL PROTECTED] wrote:
Try the Sipura SPA-2002.. at good prices from VoipSupply.com
We have been using those now with 0 problems. We remote provision
them from our office here. Once a minute (time configurable) each
device checks in with us to check out its configuration
On 2/22/06, Darrell Long [EMAIL PROTECTED] wrote:
Correct. The XML works fine. If you need an example for the 2002, I will
see if I can strip the information directly related to our company off
and send it to you.
Hi Darrell,
I would really appreciate it if you could send me the XML file
On 2/22/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
[snip]
The information needed for XML provisioning is openly available from
sipura/linksys. The actual linksys provisioning tools may be under some
license but the XML provisioning syntax is not. It is actually
ridiculously simple.
[snip]
On 2/8/06, Arne Morten Johansen [EMAIL PROTECTED] wrote:
Oh. So how can I do this?
If I write something in PHP, how do I make it output to an Asterisk
variabel? I need to set a variable in asterisk to TRUE or FALSE based on the
result of the PHP-script.
You can find the answer to this
On 2/6/06, Mark Phillips [EMAIL PROTECTED] wrote:
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and so
is
On 2/6/06, Joseph Tanner [EMAIL PROTECTED] wrote:
well I've heard that there are open source IP phones given away for free
in WALMART, I'm seriously thinking to get couple of 'em!!
What phone would this be? I didn't notice any, but there's 5-6
Wal-Marts within an hour's drive, I'd love to
On 2/6/06, Joseph Tanner [EMAIL PROTECTED] wrote:
Funny funny. In this day of free (after rebate) PAP2s, a free (again,
I assumed after rebate) IP phone seemed plausible. BTW, check
walmart.com, they do indeed sell ip phones.
I guess I'll just have to use one of my free DTA310s or my free
On 1/13/06, Mike Fedyk [EMAIL PROTECTED] wrote:
[..snip..]
Also an ugly hack would be to call the perl bytecode instead of the text
script. That would allow for the ease of AGI (everything is cleaned up
when the process exits) with lower overhead.
FastAGI is of course what you want for
On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote:
According to the Asterisk wiki, adding the delete=yes option to a
voicemail definition should automatically delete messages after they
are emailed. This is the format that I'm using:
101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes
Hi there.
I'm having a strange issue with the distinctive ring detection in
Asterisk (I have a FXO card).
It certainly seems to be enabled as I can see the Asterisk console
spitting out the cadences (same cadence every time: 0,0,0) but the
problem is that it is not waiting 2 seconds after
On 9/28/05, Michael Blood [EMAIL PROTECTED] wrote:
Anybody ever run into a case where the Sipura Dial Plan will not work with
the S0 option to immediately connect?
My Dial plan reads
(*xx|[3469]11S0|0|00|[2-9]xxS0|1xxx[2-9]xxS0)
and I can dial ONLY then numbers in the dial plan
On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote:
I hope someone can offer me some help with this. Basically, the current CVS
version of Zaptel will not compile under Fedora Core 4. I have closely
followed the directions in
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
On 7/5/05, Dana Olson [EMAIL PROTECTED] wrote:
I think they were hoping that the client would connect to Asterisk,
which makes it kinda useless, really.. But connecting Asterisk to the
Gizmo network is handy.
Given that it's a fairly new program, we have to wait a while before
it's mature
On 5/29/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, May 28, 2005 at 06:34:23AM +1000, Gonzalo Servat wrote:
[snip]
If Asterisk allowed me to configure up to 10 ringing patterns, I could
probably cover most of the ringing patterns being detected, but
unfortunately there is a limit
Hi All,
I've recently got a second number installed on my PSTN line,
trusting the Asterisk distinctive ring detection would work as
expected. It appeared to work fine at the start, as the second number
generated a different ring pattern to 0,0,0 (in the console) only to
realise that almost every
On 5/28/05, Jay Milk [EMAIL PROTECTED] wrote:
Could you configure your normal ring to be recognized as a distinctive
ring and go into a different context? That would essentially allow you
to distinguish between the calls.
Excellent suggestion Jay! Thanks!! I changed the default context (if
no
On Apr 6, 2005 10:34 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Anyone have any ideas on where I can find the right kernel
source? I have look at rpmfind.net and
google'd with no avail!Hi,
You're never going to find the kernel source. The reason for this is
that your VPS
On Apr 8, 2005 12:48 PM, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Call Accounting is such an important issue for me it is literally a make orbreak component, without it I will not be able to deploy Asterisk at ourresort. If I have to use a windows computer to download and run the clientend of
Hi,
Has anyone else experienced problems as of the last couple of months
when outbound calling through Freshtel?
I've started getting a No authority found error. I've tried
contacting them, and they seem to have some serious communication
issues with their IT team, infact I think they have
On Thu, 17 Feb 2005 10:29:54 +, Alistair Cunningham
[EMAIL PROTECTED] wrote:
I've been considering doing a web based database system, where you can
post your termination offerings or wanted, then search by location,
price, minimum volumes, etc.
(snip)
Great idea Alistair. Would certainly
On Tue, 2004-11-23 at 19:06 -0600, Joe Greco wrote:
[..snip..]
On the flip side, senders of spam should not expect recipients to go
to much (or any) trouble on their behalf, especially given the current
spam environment on the 'net. They - not hackerwaCker - blew the
surprise by sending
On Tue, 2004-10-19 at 16:56 -0500, Matthew Boehm wrote:
No. The link you gave is for a PAP2 NOT a PAP2NA. There is a HUGE OH MY GOD
difference between the two model numbers.
What is this huge OH MY GOD difference between the two? (apart from the
-NA). I've googled and can't seem to find any
Hi Stewart,
Nice project! Something I'd certainly love to be doing myself. Anyway,
the following replies I've made to your questions are based on my
experience and past research. There may be better/cheaper alternatives.
In any case, I hope it helps:
On Fri, 2004-10-15 at 12:05 -0400, Stewart M.
Salutations,
In hopes of accelerating the adoption of Asterisk and changing the
landscape of the small business marketplace, we are contributing our
administration interface to a new project that aims to bundle
best-of-breed applications to produce a canned (but fully
On Wed, 2004-09-15 at 16:10 +0200, wrote:
Hello!
I have been googling a lot and asked wiki a few times now, but i cant find
a howto for setting up a voicebox.
Any link/hint would be great!
I'd hate to refer you to the Wiki but the answer is in there :) (you did mean a
voicemail box, right?)
Correcto, I think it's also In My Humble Opinion too.
Gonzalo
P/D: Como andas Seba... :)
On Wed, 2004-07-14 at 11:45 -0300, Sebastian Nocetti wrote:
IN MY HONEST OPINION... IMHO
I am right?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
On 9/07/2004 10:21 AM +0700, Isianto Istiadi wrote:
Dear guys,
I'm searching the wake-up call script in wiki, found one, but I have no
idea how to use it. Can you give some direction how to install it?
Thanks
I presume you're talking about this wake up call script:
On 9/07/2004 6:06 AM +0100, Kevin Walsh wrote:
Eugen Cristea [EMAIL PROTECTED] wrote:
Find local movie times and trailers on Yahoo! Movies.
http://au.movies.yahoo.com
What does Yahoo have to do with it?
Have you considered trimming your quotes? Clearly not.
Have you considered maybe his webmail
On Sat, 2004-06-26 at 10:09 +0100, Dee Lowndes wrote:
Hi all,
Anyone know where/how I can setup my own menu to work like the
voicemailmain menu.
e.g.
extension.conf
exten = 888,1,mymenusystem
exten = 888,2,Goto(s,6)
then somewhere mymenusystem plays message and give options
On Thu, 2004-06-17 at 09:21 -0400, Troy Settle wrote:
..snip..
However, my preference is for top posting. The reason, is that in order to
read my message here, you had to scroll through ~70 lines of previous
discussion. Stuff that you've /already/ read since you've been following
this
On Thu, 2004-06-17 at 08:20 -0700, Deepak Malhotra wrote:
Hello
Any idea or code on How to allow users to change their voice mail
password over the Phone.
The only way io know is to change in voicemail.conf file and restart
asterisk.
Try dialing your voicemail extension, enter your
Hi All,
Whenever a call comes in via the ISDN and somebody leaves a voicemail,
the sound file recorded is very choppy. If I actually take the call, the
sound is not choppy so it's obviously something to do with the Asterisk
box itself having to do the recording. Perhaps the sound card drivers?
On Thu, 2004-06-10 at 16:27 +0100, Simon wrote:
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
The rumours are true! You would look in the ever-so-helpful Wiki:
Hi All,
I'm running the latest asterisk CVS code (from experimental), and
hardware-wise an AVM Fritz!PCI Card, together with chan_capi 0.3.3.
The problem is that any inbound/outbound calls result in echo on MY end
(the asterisk end). I've played with the echo settings in capi.conf
(mainly
On Wed, 2004-05-26 at 14:42 +1000, Simon Brown wrote:
Can I do this with * ???
S,1,answer call
S,2,play thanks for calling, we'll be with you soon
S,3,play music while caller waits and ring nominated extensions at same time
S,101,if not answered go to voicemail
I can't find a way to play
On Mon, 2004-05-24 at 09:57 +1000, Andrew Yager wrote:
Thanks! That's good to know. Please excuse my ignorance - if we have
two telstra ISDN2 lines, which card should I get?
A somewhat reasonably priced ISDN card that works with Asterisk and is
sold in Australia is the AVM Fritz:
On Mon, 2004-02-02 at 23:19, jjj3 jjj3 wrote:
I'm trying to compile the last * CVS version and I got this error:
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
-O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
Hi,
On Tue, 2003-12-23 at 12:12, [EMAIL PROTECTED] wrote:
[...]
I am trying to compile the asterisk and if fails at the end
on:
make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx'
gcc -shared -Xlinker -x -o pbx_gtkconsole.so
On Tue, 2003-12-16 at 10:34, Michiel Betel wrote:
Is case anyone wants to know... The Fritz! USB ISDN box works fine with
Asterisk!
I'm running CAPI 0.3.0 and love it, because the mini ITX server I have
only takes one PCI slot which is now filled with a 4 port Digium card.
Hi Michiel,
On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote:
Yes, echo problems do still exist, I would suggest testing it before
going live.
Yeah, so I've heard.
A couple of points to note:
1) Using soft phones seems to compound the issue
So the echo problems are not so bad when using software
them? I have found them to be most helpful with any problems (mainly
with the Pulsar PCI ADSL cards)
Try talking to [EMAIL PROTECTED] ?
-Bryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gonzalo Servat
Sent: Tuesday, 18 November
person contacts him then all of
us:)
Thanks,
Matthew Enger
[EMAIL PROTECTED]
On Tue, 2003-11-18 at 01:10, Gonzalo Servat wrote:
I'll be speaking to Guy tomorrow about this. Guy is certainly a helpful
friendly guy and I'm sure he'll be keen to hear about these echo
problems
Hi All,
This topic has come up before in the Asterisk mailing list many times,
so I know that a lot of people have given up in waiting for a FXO card
to be approved by the Australian telecommunications authority. My
question is: all legalities aside - is anyone using a FXO card in
Australia
On Mon, 2003-11-17 at 12:20, Anthony Wood wrote:
I have spoken to a number of Australian users who are successfully using:
X100P
NetJet (echo issues)
AVM Fritz!Card
I hope to add myself to their number shortly, since we have recieved our Fritz!es
Also [EMAIL PROTECTED] seems to be
Hi All,
I was wondering what the status of distinctive ring support in Asterisk
is? I had a google search read and Mark Spencer wrote some support for
it.
Is distinctive ring different in every country or is it pretty standard?
And for my final question, does the Wildcard FXO card support
On Mon, 2003-11-17 at 16:00, Anthony Wood wrote:
ISDN (telstra Onramp 2) is very similar in price to standard telstra lines.
The only problem is you can't have ADSL ISDN on the same line.
We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250.
I was a bit turned off by
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