Thanks, I will look into them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Saturday, July 02, 2005 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Provider Survey
List
Hello,
My question is about which Linux distribution to choose for
Asterisk. (/me holds breath) OK, hopefully youre still reading,
because whatever you were thinking now, youre thinking wrong! ;)
First of all, I want to make clear that I have read EVERY
message and reply that I could
On Jul 2, 2005, at 1:00 PM, Roland Zagler wrote:
sorry for the misunderstanding, robert!
the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played
Jay Milk wrote:
asterisk -nr
n - no colors
I don't understand this answer. Below is his question again:
when I start asterisk directly, I get a colored CLI. When
connect to a
already running asterisk with asterisk -R, it's never
colored, despite
I'm running both from the same
On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote:
Robert Goodyear wrote:
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I try
to dial an extension (from X-Lite), next
Hi TWV,
My question is about which Linux distribution to choose for Asterisk.
(/me holds breath) OK, hopefully you’re still reading, because
whatever you were thinking now, you’re thinking wrong! ;)
Still, my question was not answered! Mainly because the same answer
always came back: “Use
I am interested. I'm using Callvantage and Centillium TA. I'm also using
Asterisk with a X100P Card. Eventually I want to replace this with my
Linksys NSLU2. I'm having mystery power issues with my linux server.
Currently I have calls from soft and ip phones going to asterisk then out to
Hi
On Sun, Jul 03, 2005 at 08:45:05AM +0200, TWV wrote:
Hello,
Disclaimer: I'm a Debian fan and also maintain a small Debian derivative
distro specilized for Linux.
My question is about which Linux distribution to choose for Asterisk. (/me
holds breath) OK, hopefully you're still
Hello!
Thank you for the hit :)
Do you know if it works well with asterisk or not?
Thanks,
Tamas
Andrew Latham wrote:
Just get one of these.
The PCI 921-CDS is a low-cost channelized DS3 WAN adapter that can be
used in ImageStream's Industrial Series routers or OEM products
Hi the list !
I totaly agree Tzafrir.
I am an happy Debian user from a while now.
I recommand to my friends to use Testing branch to have the latest packages
(which are stable or near to be stable but usable).
Asterisk is delivered closed to be complete Asterisk
1.0.7-BRIstuffed-0.2.0-RC7k with
On Sat, 2 Jul 2005, Michael Stahl wrote:
The system startup script /etc/init.d/asterisk calls the script
/usr/sbin/safe_asterisk
In safe_asterisk, the program is started with -c by default (console on
TTY9).
That explains why it is starting with a console, but not why it's
running so
On Sun, Jul 03, 2005 at 11:01:44AM +0200, [EMAIL PROTECTED] wrote:
Hi the list !
I totaly agree Tzafrir.
I am an happy Debian user from a while now.
I recommand to my friends to use Testing branch to have the latest packages
(which are stable or near to be stable but usable).
Asterisk is
yes, robert, but how do i join the two legs inside a call file or
in the dialplan?
i have experienced that call files can do a call to a channel and
if this call is answered it can either be connected to an extension
inside a context or call an application with parameters.
roland
-Original
Hi all,
Ive got 3 analog phones and 2 grandstream sip phones working with
asterisk, the problem is that although the analog phones can talk to
each other and the sip phones can talk to each other the two types dont
seem to be able to cross communicate.
It looks as though the SIP phones are
i dont know how to edit the time 3ms for ringing
in astcc when it says there is no body to answer.i
want to change this time to 4ms but i dont know
how.please help please.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
Dear All,
I read your notes and was very glad, it was a healthy and useful debate,
I have set my mind on implementing Realtime for sipusers and peers with
mysql database and either use the Mysql replication process or mount the
database on both servers.
I will write a document of this trial and
Mohamed A. Gombolaty wrote:
Dear All,
I read your notes and was very glad, it was a healthy and useful debate,
I have set my mind on implementing Realtime for sipusers and peers with
mysql database and either use the Mysql replication process or mount the
database on both servers.
I will
Tzafrir Cohen wrote:
On Fri, Jul 01, 2005 at 09:53:33AM -0700, Wiley Siler wrote:
Anyone know a good distro for an Epia Mobo with the C3 chip?
Debian, as for any hardware :-p
Heya,
I have heard that epia C3 has full i586-support, but i686 support is not
complete.
greetings,
so here it is, the problem that's been nagging me for the past 2 days:
connected a box to my telco's NTBA - zap/asterisk. which works:
box:/etc/asterisk# cat /proc/zaptel/1
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7) HDB3/CCS
1 ZTHFC1/0/1 Clear (In use)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Keith Caldwell
Sent: Saturday, July 02, 2005 8:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head
Here is my configuration everything would seems be straight forward, but
I can not register both asterisk with each other.
Both asterisks have Static IP but they are behind firewall/router, so
it means I have to use Register statement.
I'm a bit confused about the register statement.
Hello,
I would recommend Slackware mostly for it's streamlined, minimalist approach
and history of stable distro releases. But with that said, the most
important thing is building a custom streamlined Linux kernel no matter what
distro you use. This can save you bootup time as well as speeding up
Wow,
I just want you to know I am and have been a
Networks Engineer for many years. I started back when Novell was king for
networks. Window and many others have come by and I have setup shop with
them. I still manager and maintain several of my Clients Windows
networks. Almost 3 years
Ive got 3 analog phones and 2 grandstream sip phones working with
asterisk, the problem is that although the analog phones can talk to
each other and the sip phones can talk to each other the two types dont
seem to be able to cross communicate.
It looks as though the SIP phones are set to
i noticed that the sound volume of the zap(tdm400p)
was low ,so i tried to raise the sound volume but i
didnt know how please help me.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hrmmm small simple easy to manage, and very clean minimal install -
FreeBSD, if you want linux Id also say Debian
On Sun, 2005-07-03 at 09:16 -0400, Ariel Batista wrote:
Wow,
I just want you to know I am and have been a Networks Engineer for
many years. I started back when Novell was
Hi:
I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that sell IP and
forward it to IAX file as in the DID service. Any
reference or recommendations please?
Regards;
Yahoo! Sports
Hi Nicolás,
Thanks for all your assistance. How are you keeping ?
Unfortunately I'm still battling with this one:
Since we've chatted the driver developer of the Sirrix board has now
configured the driver so that channel names are now displayed as:
Srx/gout--01
Srx/gout-0001-01
etc. In
Just my two cents... but with that ambitious deployment plan, you
shouldn't ask about distros, but rather start interviewing some linux
admins. If you're really planning on deploying all those machines on a
platform foreign to you, the distro is a fraction of your battle.
You'll need to
Hi:
I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that sell IP and
forward it to IAX file as in the DID service. Any
reference or recommendations please?
Only your ISP can sell you an IP. Please contact your ISP.
~j
Hello all,
I'm evaluating a VRU project which has huge requirements. I'm looking
for metrics but I haven't found anything that cover my requirements
Initial estimation:
Erlang 61.450
BTH 25.980
T1 req. 88
Digium HW support 4 T1 per card, assuming
Hi:
I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that sell IP and
forward it to IAX file as in the DID service. Any
reference or recommendations please?
Only your ISP can sell you an IP. Please contact your ISP.
Not /quite/ true. You
Hi the list,
Searching to start ztmonitor in quantitative mode rather than graphical.
I want to read the real voltage (RMS) or dBm on my analog telephone lines..
TIA
Best Regards,
Francois BERGERET,
France.
___
Asterisk-Users mailing list
My question is about which Linux distribution to choose for Asterisk. (/me
holds breath) OK,
hopefully youre still reading, because
whatever you were thinking now, youre thinking wrong! ;)
First of all, I want to make clear that I have read EVERY message and reply
that I could
I don't know about channels working but the data would
On 7/3/05, Tamas J [EMAIL PROTECTED] wrote:
Hello!
Thank you for the hit :)
Do you know if it works well with asterisk or not?
Thanks,
Tamas
Andrew Latham wrote:
Just get one of these.
The PCI 921-CDS is a
Robert,
I think you jay be wrong on this. From the /etc/asterisk/extensions.conf
file included with [EMAIL PROTECTED]:
; Customizations to this dialplan should be made in
extensions_custom.conf ; See extensions_custom.conf.sample for an
example #include extensions_custom.conf
Tom
chawki hammoud wrote:
Hi:
I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that sell IP and
forward it to IAX file as in the DID service. Any
reference or recommendations please?
I suspect what you are really asking for is a way to gateway a DID to
Hi
As a beginner of Asterisk server I still have some problems running CallerID on
ordinary analog Swedish phones.
I use a Digium card TDM400P with 1 FXO and 3 FXS adapter cards. The Asterisk
version running is the lates 1.0.9.
I have also a Vood I3Micro ATA adapter running SIP protocol towards
Quoting TWV [EMAIL PROTECTED]:
My question is about which Linux distribution to choose for Asterisk. (/me
holds breath) OK, hopefully you're still reading, because whatever you were
thinking now, you're thinking wrong! ;)
Your email is an excellent example on a well researched question a
On Sunday 03 July 2005 02:45, TWV wrote:
Still, my question was not answered! Mainly because the same answer always
came back: Use the one you are most comfortable with. Well, I already
knew that (linux is linux), but it doesn't apply to my situation at all!
Then you don't understand or
Dear All,
I need to bind two different ports at the same time for SIP.
5060 and another port number.
Is it possible ?
It would be something like
port=5060,5062
Isamar
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have a good quantity of IP numbers for sale.
Meet me at the Brooklyn Bridge tomorrow at 14:30.
Oh yeah, you can have the bridge too, make an offer.
On Sun, 3 Jul 2005, chawki hammoud wrote:
Hi:
I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that
Hi list,
another question for you all, and i apologize in advance if it is
basic, the syntax is making me crazy and the documentation is no help:
when i do database show in the console, i get the following:
/DB(CFIM/999) : 999
and when i run the following
Hello there,
I'm trying to configure my voicemail system and I have a couple of
questions:
* Is real-time voicemail already working? If so, where is it that I
should specify the database name, user and password? Where can I get
more information about the different options that exist and
Hello everyone,
I'm a new Asterisk user and I wonder what people usually do to keep
track of who is doing what in the system. For instance, I would like to
keep track of who is calling certain contexts/extensions and when that
happens. I'm also interested in logging customized status
Hi everyone,
I wonder if people could send me sample configurations showing how to
deal with user authentication in Asterisk. Is there anyway to integrate
user authentication with voicemail passwords? Is there any central
module that handles authentication in Asterisk?
Thank you so much,
On Wed, 29 Jun 2005, Paul Fielding wrote:
I have indeed already done so - I use G729 quite a bit since I travel alot
in adverse conditions. Interesting thing is, I can get less choppy audio
sometimes from my Vonage device using (what I suspect to be) Ulaw, while
either ulaw or G729 will
I went with Fedora - great support and eas of use (because of Red Hat
shared tools).
So far so good!
-Original Message-
From: Scott Kamp [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 03, 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
[snip]
Below is an example of a working type=user and type=peer without
the register statement prior to livevoip going bankrupt.
;[livevoip] ; for incoming calls from LiveVoIP.com
;type=user ; used for Incoming calls
;secret=mysecret-in
;deny=0.0.0.0/0.0.0.0
Hello Leo,
You can use the CDR option. Log's everything to a mysql or postgresql
database and use another program to query the database. Make graphs and
reports.
For such a program check this link:
http://areski.net/asterisk-stat-v2/about.php
good luck
/* Ferdy */
info(AT)nsec(DOT)nl
Leo
On Sun, Jul 03, 2005 at 04:33:29PM +0100, Subhi S Hashwa wrote:
Quoting TWV [EMAIL PROTECTED]:
My question is about which Linux distribution to choose for Asterisk. (/me
holds breath) OK, hopefully you're still reading, because whatever you were
thinking now, you're thinking wrong! ;)
* Is real-time voicemail already working?
Yes, and has been working for quite some time. Since December I believe.
If so, where is it that I
should specify the database name, user and password?
Read docs/README.extconfig
Where can I get
more information about the different
On Sun, Jul 03, 2005 at 01:02:08PM -0400, Michael Stahl wrote:
I went with Fedora - great support and eas of use (because of Red Hat
shared tools).
So far so good!
However so far relatively short supported life. Is Fedora Legacy showing
enough signs of life to be an actual source of support?
On Sun, Jul 03, 2005 at 07:19:39PM +0200, Ferdy Riphagen wrote:
Hello Leo,
You can use the CDR option. Log's everything to a mysql or postgresql
database and use another program to query the database. Make graphs and
reports.
Or feed the default CSV file to a spreadsheet of your choice and
On Mon, Jul 04, 2005 at 01:05:01AM +0900, Isamar Maia wrote:
Dear All,
I need to bind two different ports at the same time for SIP.
5060 and another port number.
Is it possible ?
It would be something like
port=5060,5062
Why do you need it?
You can probably get something similar with
On 7/3/05, Jerry Glomph Black [EMAIL PROTECTED] wrote:
On Sun, 3 Jul 2005, chawki hammoud wrote:
Hi:
I have my Asterisk server behind a nat and I want to
buy a static IP. Is there a company that sell IP and
forward it to IAX file as in the DID service. Any
reference or
Title: Message
Hi
all,
I am currently using
the latest CVS head to do realtime and it is working just fine so
farhowever, I was asked to use a Quintum Tenor AX gateway for testing out
the SIP send/receive call abilities in Asterisk and unfortunately I have no
experience with Quintum
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
Actually this is incorrect: Everybody can provide support for CentOS,
Debian, Gentoo, Slackware, FreeBSD, or whatever. With RedHat or SuSE
you're locked to a single vendor to provide maintinance.
You're already locked in (to Digium)
You're missing my
On Sun, Jul 03, 2005 at 08:06:13PM +0100, Subhi S Hashwa wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
Actually this is incorrect: Everybody can provide support for CentOS,
Debian, Gentoo, Slackware, FreeBSD, or whatever. With RedHat or SuSE
you're locked to a single vendor to provide
On Sunday 03 July 2005 15:06, Subhi S Hashwa wrote:
In a business enviroment, you discovered an undocumented issue with XYZ I
want to have the peace of mind and assurance that I woun't be stuck out in
the cold waiting for a response from a mailing list or getting insults from
someone on irc,
I like Mandrake/Mandriva linux (cooker) - it has asterisk cvs and other
usefull stuff and addons for asterisk (icd, sccp,... ), all prepared as
rpm packages and regularly updated ;-)
PJ
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Asterisk-Users@lists.digium.com
Sell 'em quick, 'cause here comes IPv6 and something tells me the
market's going to be saturated. Hmmm... what to do with four and a
quarter billion (round numbers) addresses... I know, porn sites!
-Bryce
On Jul 3, 2005, at 11:08, Mark Charlton wrote:
On 7/3/05, Jerry Glomph Black [EMAIL
Hi,
I'm following the instructions on Asterisk Doc Proj:
astersik_1.0.8/docs/docs-html_one/vm1.html#AEN30 and they don't seem to
work out.
The order suggested is:
cd zaptel
make clean
make linux26
make install
vi Makefile (and uncomment ztdummy)
make
modprobe zaptel
modprobe ztdummy
Not /quite/ true. You can also purchase VPN service, which could be
used to tunnel through a NAT device and provide a full visibility IP
on the inside network. That's probably not the best route to go for
a VoIP solution, but then again, if the networks are well connected,
it could be just
LOL
On 7/3/05, Jerry Glomph Black [EMAIL PROTECTED] wrote:
I have a good quantity of IP numbers for sale.
Meet me at the Brooklyn Bridge tomorrow at 14:30.
Oh yeah, you can have the bridge too, make an offer.
On Sun, 3 Jul 2005, chawki hammoud wrote:
Hi:
I have my Asterisk
Hi,
I've had an inquiry for a small UK call centre, mostly outbound calls.
I get the impression they
are mainly calling 3G mobile phones, monthly phone bill, with calls is
approx £5,000 for several
feature lines.
How feasible is something like this with asterisk?
I guess one big question is
I beleive queues will do it all for you.
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+queues.conf
Use this part to play the sound to the callers:
;queue-youarenext = queue-youarenext ; (You are now first in line.)
and use:
;announce = queue-markq
Jon, etc..., the issue here is her family all uses
special features of SBC voicemail. E.g. Her mom
leaves a VM for my wife at 10:30 PM after the baby is
asleep without fear of ringing the phones and possibly
waking up the baby. They use it like email. They
The only way I see is make an
Has the Digium DS3 card been released yet?
Rob
On 7/3/05, Manuel Soto [EMAIL PROTECTED] wrote:
Hello all,
I'm evaluating a VRU project which has huge requirements. I'm looking
for metrics but I haven't found anything that cover my requirements
Initial estimation:
Erlang
Anybody wants to buy a Cuban president ?
If somebody buy it we will provide him with
256 FREE publics IP address ;o))
jat
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 03,
Mike Dent wrote:
I guess one big question is which type of circuit to use, ADSL in the
UK is only 256kbs upstream,
some providers do bonding but I'm not sure its supported fully by BT :(
The other option is SDSL which is not too cheap!
SDSL is dirt cheap. 1MB SDSL is only £275/month.. you're
[snip]
Below is an example of a working type=user and type=peer without
the register statement prior to livevoip going bankrupt.
;[livevoip] ; for incoming calls from LiveVoIP.com
;type=user ; used for Incoming calls
;secret=mysecret-in
;deny=0.0.0.0/0.0.0.0
I'd like to these three things about asterisk:
1. How the asterisk program can be configured to run as a different user from
root.
2. what directories and files it must have read and right access to
3. Setup an asterisk group, which also has some of the rights the asterisk user
has rights to,
On Sun, Jul 03, 2005 at 11:31:50PM +0100, Tony Hoyle wrote:
Mike Dent wrote:
I guess one big question is which type of circuit to use, ADSL in the
UK is only 256kbs upstream,
some providers do bonding but I'm not sure its supported fully by BT :(
The other option is SDSL which is not too
Scenario:
Both boxes are behind firewall, port udp 4569 is open.
If I don't want the username and password in dialing string do I have to
use register statement in IAX.CONF.
Can anybody post some working samples; I have a hard time making it to
work with the samples posted on wiki.
--
#Joseph
Hi Joseph,
here is how i did it:
iax.conf of server1:
[server2]
type=friend
auth=md5
username=server1
secret=secret
context=default
host=dynamic
defaultip=public ip of server2
deny=0.0.0.0/0.0.0.0
permit=public ip of server2/255.255.255.255
disallow=all
allow=g729,gsm
iax.conf of server2:
Thanks for the suggestion, C F, but the problem is there is a rather big
database application behind with many users, so a static configuration
is not suitable for my needs. i am working mostly with realtime and agi.
regards,
roland
-Original Message-
From: [EMAIL PROTECTED]
I don't see why this doesn't work with realtime. The same it works
with .conf files
On 7/3/05, Roland Zagler [EMAIL PROTECTED] wrote:
Thanks for the suggestion, C F, but the problem is there is a rather big
database application behind with many users, so a static configuration
is not suitable
On Sun, 2005-07-03 at 17:36 -0600, Rich Adamson wrote:
[snip]
Below is an example of a working type=user and type=peer without
the register statement prior to livevoip going bankrupt.
;[livevoip] ; for incoming calls from LiveVoIP.com
;type=user ; used for Incoming calls
Mike Dent wrote:
Hi,
I've had an inquiry for a small UK call centre, mostly outbound calls.
I get the impression they
are mainly calling 3G mobile phones, monthly phone bill, with calls is
approx £5,000 for several
feature lines.
How feasible is something like this with asterisk?
I guess one
Michel Brabants wrote:
Tzafrir Cohen wrote:
On Fri, Jul 01, 2005 at 09:53:33AM -0700, Wiley Siler wrote:
Anyone know a good distro for an Epia Mobo with the C3 chip?
Debian, as for any hardware :-p
Heya,
I have heard that epia C3 has full i586-support, but i686 support is not
On Mon, 2005-07-04 at 01:09 +0200, Roland Zagler wrote:
Hi Joseph,
here is how i did it:
iax.conf of server1:
[server2]
type=friend
auth=md5
username=server1
secret=secret
context=default
host=dynamic
defaultip=public ip of server2
deny=0.0.0.0/0.0.0.0
permit=public ip of
Hello,
I'm trying to use the GSM codec with Nufone H323 but it's not working.
Does somebody has some idea? Have I missed something?
Thanks!!
Celso Fassoni
Some additional info:
(I'm using CVS-HEAD - downloaded today)
monkey:~# cat /etc/asterisk/h323.conf
[general]
port = 1720
bindaddr =
Mike Dent wrote:
Hi,
I've had an inquiry for a small UK call centre, mostly outbound calls.
I get the impression they
are mainly calling 3G mobile phones, monthly phone bill, with calls is
approx £5,000 for several
feature lines.
How feasible is something like this with asterisk?
For calls
You're leaving the :1 in the dial expression, which cuts off the
first digit so what's really being dialed to the server is only 88.
-Bryce
On Jul 3, 2005, at 19:01, Joseph wrote:
On Mon, 2005-07-04 at 01:09 +0200, Roland Zagler wrote:
Hi Joseph,
here is how i did it:
iax.conf of
Robert Goodyear wrote:
On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote:
Robert Goodyear wrote:
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an
Hello,
I am trying the setup the TDM01B card. 1 FXO. I
connected it to a regular home line. in the
/etc/zaptel.conf, I have
fxsls=4
In the /etc/asterisk/zapata.conf
I have:
signaling=fxs_ls
language=en
group=1
context=default
channel = 4
When I start asterisk, I get this error:
ERROR[10376]:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server.
Write some more about your project,
What it suppose to do ?
On 7/3/05, Manuel Soto [EMAIL PROTECTED] wrote:
I'm evaluating a VRU project which has huge requirements. I'm
looking
for metrics but I haven't found anything that cover my requirements
regards
m.
the point is: during the caller is listening to the soundfile played
to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played the caller should
be
connected
to the sip phone 100.
Ah, now that's a clearer picture of
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the 10 minute guide, located here:
Well, I found a WiFI phone that looks identical to the Hop-On one. It
is from www.broad-tel.com/index_en.php but is selling at $180/each for
every 20 units.
On 6/29/05, William Suffill [EMAIL PROTECTED] wrote:
Unfortunately no. Someone say the press release and bugged me about it
as well but I
Thanks Bryce, I got it, it took me several hours :-/
At least I got the two asterisk servers working, that is the good
part :-)
--
#Joseph
On Sun, 2005-07-03 at 20:15 -0700, Bryce Chidester wrote:
You're leaving the :1 in the dial expression, which cuts off the
first digit so what's really
93 matches
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