hello, my experience
1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE
2.- MTA-V102
3.- Sipura spa 2000
4.- Granstream
ATA186 SUXs
Excuse me I have just bought a PAP2 ,, is it true that only one g729,
one of the Damn things Cisco had in the ATA186? at the same time.
DAMN , its just a Sipura
Hello,
I have an Asterisk server. When I connect
to the console (asterisk -r) and I want to see the time that the server
has been connected (CLI show uptime) I noticed that Asterisk restarts
alone. Why?
Any clue will be apreciated. Best Regards,
Hi,
I am mostly happy with my Polycom IP600 but it apparently needs to check
the FTP server every minute. I couldn't find any obvious setting related
to that behavior in the configuration files.
Any idea how to curb the IP600's spurious network activity?
Thanks,
--
Lord, protect me from your
Hi list!
I have some sip phones and Sipura ATA 2000's. However after dialling a
number I need to dial a # to control a device.
When I dial # Asterisk kicks in and puts the call on hold. How can I
change this?
Thx!!
Remco
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On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote:
Hi,
I am mostly happy with my Polycom IP600 but it apparently needs to check
the FTP server every minute. I couldn't find any obvious setting related
to that behavior in the configuration files.
Any idea how to curb the
On Tue, Feb 15, 2005 at 07:56:10PM +1100, Adam Goryachev wrote:
On Tue, 2005-02-15 at 09:38 +0100, Louis-David Mitterrand wrote:
Hi,
I am mostly happy with my Polycom IP600 but it apparently needs to check
the FTP server every minute. I couldn't find any obvious setting related
to that
On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote:
You are right, this activity is related to logging.
After consulting the admin manual I am unsure as to what settings
related to logging are safe to change (some are marked as don't modify
without consulting Polycom).
Do
Hi guys,
I haven't had the opportunity to play with any Polycom products,
although they will probably be the best IP phone available.
I have used the Grandstream BT-101/102, the HOP-1003 (upgraded 1002)
and Zyxel telephone adapters.
My recommendation out of the tried ones would be the
Hi,
I have following problem. Asterisk is connected to ISDN router on BRI
interface. ISDN PBX is connected to another channel of BRI interface. Now
I'd like to route all incoming calls first to Asterisk and then if caller
wants to talk to extension on ISDN PBX then I'd like to route call to
On 15 Feb 2005, at 05:44, Rod Bacon wrote:
Some more info on my problem that someone may be able to explain.
The debug information (shown below), lists the LENGTH of the CallerID
string
as 14 characters, even though I'm only sending 10. I belive that this
is the
problem. My telco's equipment is
On Tue, 15 Feb 2005 10:45:16 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I have following problem. Asterisk is connected to ISDN router on BRI
interface. ISDN PBX is connected to another channel of BRI interface. Now
I'd like to route all incoming calls first to Asterisk and then if
On Tue, Feb 15, 2005 at 08:26:42PM +1100, Adam Goryachev wrote:
On Tue, 2005-02-15 at 10:14 +0100, Louis-David Mitterrand wrote:
You are right, this activity is related to logging.
After consulting the admin manual I am unsure as to what settings
related to logging are safe to change
Hello
Any one using asterisk-prepaid with mysql. i want
asteirsk-prepaid for fedora core 2. i have installed
mysql-devel. but after that i am unable to compile the
asterisk-prepaid it is giving me error for
libmysqlclient. i already have this library in my
/usr/lib/mysql. i am using asterisk-CVS.
Robert Rozman wrote:
I have following problem. Asterisk is connected to ISDN router on BRI
interface. ISDN PBX is connected to another channel of BRI interface. Now
I'd like to route all incoming calls first to Asterisk and then if caller
wants to talk to extension on ISDN PBX then I'd like to
Hi there
I just wanted to know what the difference between [EMAIL PROTECTED] and manually
built boxes actually is ?? What makes [EMAIL PROTECTED] a home system ? Is it
not a good idea to run [EMAIL PROTECTED] then modify/tweak it to use in a
production
environment ??, if so why not, would
Hi together,
I have a asterisk running on a Debian testing system running
flawlessly at least after starting the asterisk.
The Server its running on has a fixed IP, no NAT, whatsoever and is
reachable all the time. The Firewall has holes on port 5060 and for the
RTP-range that asterisk is
hi
It looks interesting, but it is documented to support only old RedHat
versions and they don't release source to let me recompile. I am not a
big RedHat fan, but if I have to use it on the desktop, I would want
something newer than RedHat 9. If you can tell me you are using it with
a
On Tue, 15 Feb 2005, tim panton wrote:
My best advice is to call your PTT and ask them how many digits
they expect you to send, I am guessing they only expect the
last 2, but only they know for sure.
Also ask them if they require a specific Type Of Number for the outgoing
callerid.
I own a ME600 EPIA Mini-ITX main board with the latest Debian distro
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2,
isdnactivecards installed. I have a QuadBRI module by Junghanns with
bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and
libpri-1.0.3), and
On 11:52, Tue 15 Feb 05, A. Peverelli wrote:
I own a ME600 EPIA Mini-ITX main board with the latest Debian distro
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2,
isdnactivecards installed. I have a QuadBRI module by Junghanns with
bristuff-0.2.0-RC3a (with asterisk-1.0.3,
Are you running asterisk as user asterisk ?
If so, you need to add this user to the dialout group.
Otherwise it won't have access to the modem.
hope this helps.
I'm running asterisk with user 'root'. Asterisk user is in the dialout
group and I try to start asterisk as user asterisk, with the
Hi all,
I'm currently looking for a VoIP platform to support the following features:
Caller ID
Call Waiting with caller ID
Call Hold/Retrieve
Three-way conference
Calling Line Identity Presentation
Call back last missed call
Last called number redial
User line locking/Call Barring (all current
I've been thinking of making a (mostly) solid-state asterisk pbx.
Take either centos or some other distro, cut it down to bare minimum and
put asterisk + AMP on. Something that could be put onto a usb2.0 flash
stick, bootable.
Modern flash devices (usb, compactflash) have builtin wear leveling
A. Peverelli wrote:
I own a ME600 EPIA Mini-ITX main board with the latest Debian distro
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2,
isdnactivecards installed. I have a QuadBRI module by Junghanns with
bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and
Hello,
Chan_capi can be used by a billion pci card S0? So i can
fax througt it.
Thank´s
Em Fri, 11 Feb 2005 14:58:31 +0100
Stefan Gofferje [EMAIL PROTECTED] escreveu:
Anabela Abreu schrieb:
Hello, list a have a problem i can start asterisk, i
get
the fowlling error:
[chan_capi.so] =
Hello all,
I have an asterisk 1.0.3 stable instaled on a box.
All works fine with this machine, but the only problem i get is that
suddenly the machine hangs up all the establised calls and we have to
call again.
This problem occurs twice a day and i don not know how to debug it.
I read
On Tue, 15 Feb 2005, quoth [EMAIL PROTECTED]:
I've been thinking of making a (mostly) solid-state asterisk pbx.
Take either centos or some other distro, cut it down to bare minimum and
put asterisk + AMP on. Something that could be put onto a usb2.0 flash
stick, bootable.
Anyone done
Good day all
Is there any time of VOIP/SIP/asterisk qualifications or certificates?
Thanks
Altus
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To UNSUBSCRIBE or update options
I just got the latest update from the 1.0 CVS tree this morning. I was able
to make the zaptel drivers just fine, but in the asterisk directory, make
just sits there.
This is under the 2.4 kernel on a SuSE system which has worked just fine until
now.
I'm making as root, so it's not likely a
hi
the norwegian company nextgentel uses custom ATAs. does anyone know
these by view?
http://www.nextgentel.no/ressurser/brukerveiledninger/NextPhone.pdf
thanks
roy
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follow the thread.. should give you some info
- Original Message -
From: Matt Kemner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Tuesday,
Hello all,
I have an asterisk 1.0.3 stable instaled on a box.
All works fine with this machine, but the only problem i get is that
suddenly the machine hangs up all the establised calls and we have to
call again.
This problem occurs twice a day and i don not know how to debug it.
I read carefully
Hi,
I wonder what makes the difference between inserting 4 HFC-S cards (cca. 120
EUR) and using 1 QuadBRI card (approx. 700 EUR) ?
What makes such difference ? Is it possible to do first configuration ?
With what drivers ? Is it stable ?
Thanks in advance,
regards,
Rob.
With the following program:
#!/bin/sh
# mailfax: program to email received fax as pdf
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
FAXID=`basename $1|cut -d . -f1,2`.pdf
FAXTXT=`basename $1|cut -d . -f1,2`.txt
tiff2pdf $FAXFILE $FAXID
sendfax.pl $FAXID $RECIPIENT $FAXSENDER $FAXFILE
#end of program
If I
http://www.voip-info.org/wiki-Asterisk+Embedded+Systems
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Tuesday, February 15, 2005 1:44 PM
Subject: [Asterisk-Users] solid-state asterisk pbx?
I've been thinking of making a (mostly) solid-state
Hi Dan,
I've been investigating the same thing. Try to Google for Asterisk+Soekris,
Soekris is the company (http://www.soekris.com) that makes cute little 586
class fan-less single board computers that run both Linux and FreeBSD ...
Good luck,
Hans
-Original Message-
From: [EMAIL
Hi,
Somebody already made call recognition with database access?
Depending of call's number, it access a database looking for that number.
Where can i find something about this?
Thanks in advance
Pablo Fernandes
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Michael,
Someone may know a simple fix. If not, can you please install the
'strace' program, then run:
strace -f -o /tmp/strace.out make
This will run make, and log any system calls it makes to
/tmp/strace.out. When it hangs, take a look in that file. It may have
stopped on one system call,
I have had this same problem. The only way I know is to disable
transfers in asterisk. You can still use the transfer control in your
SIP device. Of course this does not work with call parking. I would
be very interested in a solution that does not require disabling of
transfers in asterisk as
Title: Fail to detect DTMF over direct ISDN pri link
Hello,
I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri.
For develop the exten context script I was using a real PSTN ISDN Megalink
Geir,
Many of your items, such as Voicemail, are not supported by SER
directly. It sounds, at least at this very early stage, as though you'd
be better off with Asterisk as it supports all of these features, though
perhaps with some development work. If need be, SER could front it for
call
Good day all
Can asterisk connect h323 clients to each other and h323 to sip and what
about h323 video?
Please Help and advice
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To
Remco Barende wrote:
Hi list!
I have some sip phones and Sipura ATA 2000's. However after dialling a
number I need to dial a # to control a device.
When I dial # Asterisk kicks in and puts the call on hold. How can I
change this?
Do you have the T in your Dial statment? Remove the T and try it.
FYI, I didn't read your message. With hundreds of messages/day, I use
the subject line to decide whether or not to read. Whenever I get a
message with (no subject) it is an instant delete.
Also, for those of you who think you're still on a 300baud modem and
have to conserve every keystroke,
Altus,
Yes, Asterisk can do the following scenarios, amongst others:
Client -- H.323 -- Asterisk -- H.323 -- Client
Client -- H.323 -- Asterisk -- SIP -- Client
In these scenarios, it is acting as a Back To Back User Agent (BTBUA).
It can also handle video calls, though I have not used this
Wondering if someone (Steve?) can clarify something form me. I think the
recent soho fax solution? thread has mixed things up for me.
- Is it possible to get reliable fax reception using
a Zaptel FXO interface connected to a standard POTS
line and a fax machine connected to
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
On 10:21, Tue 15 Feb 05, Pablo Fernandes wrote:
Hi,
Somebody already made call recognition with database access?
Depending of call's number, it access a database looking for that number.
Where can i find something about this?
You can do this with an agi script.
It's not that hard to do,
How long does it take to get
a vanity number? I signed up for an account, pre-paid some money, and then
placed a vanity number order. I did all of that around Dec. 31st 2004. They
said it would take 2-10 business days. It is now Feb. 15th and still no vanity
number. I've called them about
On Tue, February 15, 2005 7:48 am, Rich Adamson said:
2) simply switching a fax call through * to a tip/ring interface of
some sort that has an attached traditional fax machine.
Does the codec issue with #2 still apply if the incoming fax call is on a
Zaptel FXO interface? Is the codec
Sorry for posting before without a subject. Glad
to see there are those on the list who do not make mistakes. Diversity
keeps things interesting I guess.
I have a question
for using gastman. I have set up extensions for my IAX users as
IAX2/username, and I keep getting the following
On Tue, February 15, 2005 9:27 am, Rob Risner said:
I'm just wondering, how long should a vanity number transfer really take?
No help here, just posting a me too to warn others. Friday was 10 days
for me. No happy to hear you've waited much longer with the same result.
Can never raise them on
Using CVS HEAD (20050214 with the new jitter buffer) and the latest (0.0.6?)
spandsp. libtiff version is 3.5.7, compiled from source. System is
Slackware 10, 2.4.26 kernel, no fancy patches and processor is a P4 1.5GHz on
an Intel motherboard.
Most faxes are coming through fine but a few
Where can I get E1 and/or Euro-ISDN specifications/data sheets?
Are there specs for other E./G./Q./etc. protocols as well?
Thanks!
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Does anyone know of a AGI script that takes advantage of the
weather sound files thats included with the extra sound files available
from www.loligo.com/asterisk/sounds/
?
Thank,
Jeramie
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On Tue, Feb 15, 2005 at 01:16:16PM +, Alistair Cunningham wrote:
Michael,
Someone may know a simple fix. If not, can you please install the
'strace' program, then run:
strace -f -o /tmp/strace.out make
This will run make, and log any system calls it makes to
/tmp/strace.out. When
Hello,
i using asterisk with DIVA server by CAPI termination. But when i call on
off mobile phone, i can listen normaly tone, not operator message about
availability user.
Can you explain me where are possible mistake?
Thanks
___
Asterisk-Users
Rob Risner wrote:
I'm just wondering, how long should a vanity number transfer really take?
Were you requesting a new vanity number, or a transfer of an existing
number?
If it's new, have you checked to see if the number is still listed as
available?
google for vanity toll free number search
On Tue, 15 Feb 2005, Daniel Nyström wrote:
Where can I get E1 and/or Euro-ISDN specifications/data sheets?
Are there specs for other E./G./Q./etc. protocols as well?
The specifications are built one on top of another. Each just lists the
changes and clarifications relative to the underlaying
Hi all,
I use a GSM device to send dtmf on my asterisk system (via SIP).
the codec I use is ulaw (or a-law).
dtmf mode is INBAND.
relaxmode is on.
but most of the case, I 'missed' some DTMF or
I 'double' one.
as anybody as seen this before?
is there any way to prevent this
thanks
Same boat here.
Actually got someone on AOL instant messenger yesterday. Their
response as follows when asked how long it will take to get our 800
number:
[15:11] sixtel9: it's in the works
any time frame?
[15:14] sixtel9: not specifically, we switched carriers so we're
dealing w/ some issues
On Tue, February 15, 2005 7:48 am, Rich Adamson said:
2) simply switching a fax call through * to a tip/ring interface of
some sort that has an attached traditional fax machine.
Does the codec issue with #2 still apply if the incoming fax call is on a
Zaptel FXO interface? Is the
I'm trying to connect an asterisk server via oh323 to a Lucent iMerge.
I patched the code due so that Lucent can handle asterisk's ver4 h323
http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration
I can now successfully dial in to our company over multiple lines.
The issue is when I
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but still keep the T in
the dial plan? We have clients that need to check external voicemail
systems that require the use of the # sign, but still want to have the
call parking
On Feb 14, 2005, at 1:25 PM, Pedro wrote:
Is it just a bad implementation of g729 compression with the Sipura
product line?
That would be my guess.
-mark
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
Using CVS HEAD (20050214 with the new jitter buffer) and the latest (0.0.6?)
spandsp. libtiff version is 3.5.7, compiled from source. System is
Slackware 10, 2.4.26 kernel, no fancy patches and processor is a P4 1.5GHz on
an Intel motherboard.
Most faxes are coming through fine but a
On Feb 14, 2005, at 5:27 PM, Cory Andrews wrote:
There is just a form that needs to be completed, which we forward on
to Linksys and they approve or deny the application based upon the
background of the applicant. Have had very few applications rejected,
pretty straightforward process.
I
I've had the same experience. I've been waiting 7+ business days for
their unlimited incoming minutes DIDs which were supposed to be
provisioned within 1-4 hours.
On Tue, 15 Feb 2005 09:41:12 -0500 (EST), Paul Dugas
[EMAIL PROTECTED] wrote:
On Tue, February 15, 2005 9:27 am, Rob Risner said:
On Feb 15, 2005, at 3:17 AM, Voip Business wrote:
hello, my experience
1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE
2.- MTA-V102
3.- Sipura spa 2000
4.- Granstream
ATA186 SUXs
I can't speak so fondly of the Azatel which I had sitting around after
a canceling a VOIP service. Maybe I just need a
On Feb 15, 2005, at 10:09 AM, Florian Lefeuvre wrote:
Hi all,
I use a GSM device to send dtmf on my asterisk system (via SIP).
the codec I use is ulaw (or a-law).
dtmf mode is INBAND.
relaxmode is on.
but most of the case, I 'missed' some DTMF or
I 'double' one.
as anybody as seen this before?
is
Mark Eissler wrote:
On Feb 14, 2005, at 5:27 PM, Cory Andrews wrote:
There is just a form that needs to be completed, which we forward on
to Linksys and they approve or deny the application based upon the
background of the applicant. Have had very few applications
rejected, pretty
On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote:
I've had the same experience. I've been waiting 7+ business days for
their unlimited incoming minutes DIDs which were supposed to be
provisioned within 1-4 hours.
Well let me tell you one thing about that, whenever a VOIP provider
runs out of DIDs
How about writing some script that works
with a free service like weather.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeramie Rentfrow
Sent: Tuesday, February 15, 2005
9:49 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Extra
sounds
I'm sitting in a hotel close to the Madrid airport... Any Asterisk users
in the neighbourhood that wants to meet me for a beer and some Asterisk
hacking this evening?
Send e-mail to me *off list*, thank you.
/O
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Thanks Stefan - works like charm.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
Sent: 15. febrúar 2005 00:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can't run AGI for outbound call
On
Peer Oliver Schmidt posde-at-theinternet.de |Asterisk/Maestro| wrote:
A. Peverelli wrote:
I own a ME600 EPIA Mini-ITX main board with the latest Debian distro
(kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2,
isdnactivecards installed. I have a QuadBRI module by Junghanns with
Maybe trivial question, but I cannot find an answer:
How to autostart Asterisk (daemon) on Slackware 10? I know that I should
put something in /etc/rc.d, but what?
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Goran Dj. wrote:
Maybe trivial question, but I cannot find an answer:
How to autostart Asterisk (daemon) on Slackware 10? I know that I should
put something in /etc/rc.d, but what?
In my /etc/rc.d/rc.local
# Put any local setup commands in here:
/sbin/ztcfg
/etc/rc.d/rc.hdlc
/usr/sbin/asterisk
Hello All, I installed [EMAIL PROTECTED] .5 last night. I was able to
configure some extensions for the house and they work fine. I just can't
make inbound and/or outbound calls. The Flash Operator Panel shows four
external icons and my new extensions.
I have four X100P and two Broadvoice sip
On February 15, 2005 10:49 am, Goran Dj. wrote:
How to autostart Asterisk (daemon) on Slackware 10? I know that I should
put something in /etc/rc.d, but what?
Something like
/usr/sbin/asterisk -g
in /etc/rc.d/rc.local would do it. You can craft up more complex things if
you like, wrap
I've been trying this for a while and I have been unable to get a
reliable connection betwen two Zaptel FXS interfaces, so the bridging
does afect data transfer.
Anybody got some tunning tips to get this to work ?
I'm using a dual PIII with a ServerWorks Chipset, two TDM cards (8xFXS)
and a
On Tue, 15 Feb 2005, Sylvain Gagnon wrote:
I'm using Asterisk (latest CVS head) to perform outbound call as
robot/testing tool for an IVR platform, with a Wildcard T100P configure as
ISDN Pri.
For develop the exten context script I was using a real PSTN ISDN Megalink
(DMS100) to reach the
[EMAIL PROTECTED] wrote:
Hi!
Maybe I have just been looking on the wrong pages but there is a
question that is very important for me. I already studied some
Demo-Dialplans and made some basic experiences with Asterisk.
But what I
need to find out is how I can handle this.
I am leaving
On Sun, 13 Feb 2005 13:39:09 -0500, Mike Chapman
[EMAIL PROTECTED] wrote:
Hi,
I am thinking of purchasing a cheap Dlink VPN for testing purposes for use
with my Asterisk box and would like to ask the list for advice on how to
pick a VPN that will work with my box. I am a newbie to both
http://advancedippipeline.com/60400413
BOULDER, Colo. -- Leading Voice over IP service provider Vonage Holdings
has complained to the Federal Communications Commission that competitors are
blocking the use of its service, according to FCC chairman Michael Powell
and others close to the company.
Sam thing here. Waiting 10+ business days for my DID. Can't get
through to them by phone and email responses take days.
These guys are worthless.
On Tue, 15 Feb 2005 10:40:44 -0500, Mark Eissler [EMAIL PROTECTED] wrote:
On Feb 15, 2005, at 10:26 AM, BJ Weschke wrote:
I've had the same
Title: RE: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri link
Thank you Peter for you reply,
I realize this problem occur because I take the CVS head (maybe a bugs get introduce), because when I rebuild using the checkout of the latest stable version (cvs checkout -r v1-0), I
On Tue, 2005-02-15 at 13:59 +, Alistair Cunningham wrote:
It can also handle video calls, though I have not used this myself.
AFAIK video only with SIP, which I didn't test myself either. With
H323 it does not work, audio only there.
Regards, Bruno.
Hello,
I'm looking for any
comments or user experiences from anyone who is using 7912G phones with
SIP. Any installation issues? Usability problems?Do the features
seem to work, etc...In short, I'm looking for your opinions on how suitable this
phone is for an asterisk implementation for
Maximiliano,
We have implemented that solution succesfully several times.
First:
Does your Panasonic support dtmf inband signaling? without that forget it.
Also you need your setup to look like this:
Outside calls ring into pbx. Pbx co lines are forwarded to a group of
extensions set as
You can always visit Slashdot for countless (useless, well, not
always) comments:
http://yro.slashdot.org/article.pl?sid=05/02/14/2352254
--Luki
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Is the Cisco phone book via XML something specific to [EMAIL PROTECTED], or is
this
something that can be implemented within a normal Asterisk deployment..?
Thanks
On Mon, 14 Feb 2005 17:43:36 -0800 (PST)
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] has Asterisk, a TFTP
BJ Weschke wrote:
I've had the same experience. I've been waiting 7+ business days for
their unlimited incoming minutes DIDs which were supposed to be
provisioned within 1-4 hours.
Did you get any notice from them on the DID?
The dropdown for unlimited use DIDs only gives a choice for Area Code.
Hey Everyone,
I downloaded and installed the X-Lite softphone the other day (the lite
version) and cannot seem to get it to work well.
Don't get me wrong, it registers with my asterisk server and everything
seems to work well, except the call quality really is horrible.
I thought it may be the
Yeah, I'd like to hear you guys' opinion instead of CleverNickName's!
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 15, 2005 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port
-Original Message-
Sorry to bump this post, but I'm losing my mind a bit on why Speex
won't negotiate for me. Is _anyone_ using it out there?
Here's my original query with the CLI log
http://lists.digium.com/pipermail/asterisk-users/2005-February/
088225.html
I hate to check the obvious,
I've used them too and got absolutely nothing from them. My e-mails
hardly ever get responses and when they do respond, it's usually a
one-liner that evades the question. Stay as far away as you can from
Sixtel / IAX.cc. I think a BBB complaint about them should be made.
Mohit.
On Tue, 15 Feb
will be nice to have this setting posted. Here in Panama we use lots
of Panasonics and that is a nice one to have
Cualquier cosa nueva me la hacen saber por este posting o a mi correo
eaperezh @ gmail.com
Saludos,
On Tue, 15 Feb 2005 13:38:26 -0300, Sergio Veltri
[EMAIL PROTECTED] wrote:
Has anyone had stability issues with IAX2. (Asterisk 1.0.5).
reddwarf*CLI iax2 show firmware
Device Version Size
iaxy 22 39344
I'm asking because in the last three weeks I've noticed the following
two issues (on separate occasions):
1) Placed a phone
Hi
while on a call.. did you check your CPU usage.. i
have a P3 and sometimes
when i move my mouse, xlite starts to stutter.. cpu
then running 100%
just my 2cents
chow
L
- Original Message -
From: Richard J. Sears [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday,
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