On Mon, 11 Oct 2004 00:10:26 +0100, David J Carter
[EMAIL PROTECTED] wrote:
I beleive Telappliant in the UK are doing them for £55, ($35)
http://www.voiptalk.org/products/index.php?cPath=27
Dave
£55 is more like US$100 :-)
___
Asterisk-Users
Simon Brown wrote:
I have just downloaded V1.0 from CVS and when I try to start Asterisk (after
compiling and installing) I get this error:
Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:6205 mkintf: Unable to get
parameters
Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:9134 setup_zap: Unable to
I have been running Asterisk happily for many months and I was trying to
upgrade from CVS-HEAD-08/13/04-10
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of el Flynn
Sent: Monday, 11 October 2004 16:09
To: Asterisk Users Mailing List -
dean collins [EMAIL PROTECTED] writes:
Lol, you're kidding right, go and look at what it costs to buy an
alternative ip-pabx in comparison, and sorry but no corporate budget
here, this is just a system for my home $100 on an old P3-700, and about
the same on a card, and 2 $55 grandstream
From what I have seen so far on this list if you are running a version of
CVS-Head prior to release of Asterisk 1.0 then you should keep it and not
try to change or upgrade it. It would appear that there are a lot of recent
changes that may break if you try to upgrade to current CVS-Head, and
There is no
USE_MYSQL_FRIENDS and USE_SIP_MYSQL_FRIENDS in .../asterisk/channels/Makefile
any more. But, on voip-info wiki it still says that it should be configured like
this. Anyone knows how should I tell Asterisk to use mysql database for SIP and
IAX friends?
Thanks
Tomica
Crnek
Hello,
this is not really much of an issue any more in Europe, the old
state-owned monopoly phone companies have had to loosen up in the face of
private competition and de-regulation (or rather, fairly liberal re-regulation).
I something I hook up causes an actual technical malfunction in the
Except that £55 is more like $75-80 and not $35.
Regards, Wolf
David J Carter [EMAIL PROTECTED] writes:
I beleive Telappliant in the UK are doing them for £55, ($35)
http://www.voiptalk.org/products/index.php?cPath=27
Dave
Grandstreams are availabe for $65 quanity one, so its not hard to believe
Jeremy McNamara wrote:
Mészáros Mihály wrote:
Please if you can please help me to solve this problem.
Help yourself and READ THE README.
Hello Jeremy!
I read it already! ;-) thx!
But i didn't find a word about that chan-h323 what decoder encoder use.
It use the libopenh323 or other in built
Hi,
I am looking for a basic script that can reload asterisk from
php or perl via a web browser.
I have tried exec( asterisk -rx reload ) and shell_exec( same cmd )
with php but there seems to be a permission issue with asterisk that
stops these working. I was just wondering if
$1.64 to the £1 I think this morning so $35 stands.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wolf N.
Paul
Sent: 11 October 2004 07:40
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream phone price
Except that £55 is more like $75-80
Forget the last post, the brain is totally screwed. Must get more sleep.
Thanks all for pointing the errors of my conversion, so used to working the
other way.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wolf N.
Paul
Sent: 11 October 2004
On Sun, 2004-10-10 at 22:32 -0500, Steven Critchfield wrote:
On Mon, 2004-10-11 at 00:10 +0100, David J Carter wrote:
I beleive Telappliant in the UK are doing them for £55, ($35)
http://www.voiptalk.org/products/index.php?cPath=27
Your conversion above is going the wrong way. a British
oi geli wrote:
Can I use the Asterisk to register to a H323
Gatekeeper as client? I have the GK IP address and the
user id. I am using chan-h323 (from CVS).
Please share the h323.conf if you have it working. I
did not see any GK user id field in the h233.conf.
Thanks
We are using * as SIP,
On Mon, 11 Oct 2004 09:23:54 +0200, Dave Cotton
[EMAIL PROTECTED] wrote:
No Steve, He works for NASA. :)
hilarious :-)
This reminds me of an anecdote I'd like to share ...
After WWII, US Army officials set new values for measurement units in
defeated Japan. At some point they came to a unit
On Mon, 11 Oct 2004 08:35:06 +0100, David J Carter
[EMAIL PROTECTED] wrote:
$1.64 to the £1 I think this morning so $35 stands.
I can only hope you are not working on any billing software ;-)
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB:
You multiply to get the dollar price.
Careful where you go on holiday, it could be costing more than you think!!
-Original Message-
From: David J Carter [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 08:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Hi
can somebody who has got an IAXy please run a debug on their Asterisk
server and send me the session transcript of an attended transfer
(assuming the IAXy supports this) ?
I am currently creating call flow charts for IAX call scenarios to
assist a phone manufacturer to implement IAX and
In article [EMAIL PROTECTED],
David J Carter [EMAIL PROTECTED] wrote:
$1.64 to the £1 I think this morning so $35 stands.
Dave
So that makes £55 to be 55 x $1.64 = $90.20
It wasn't the rate that was at issue, but that the OP divided instead
of multiplying.
Cheers
Tony
--
Tony Mountifield
Hi Robert;
First, you have to use the SIP2 channel code (chan_sip2.c) from
http://bugs.digium.com/bug_view_page.php?bug_id=759 as this does the
proxy-authenticate properly.
Get the module, follow the build instructions, and add noload=chan_sip.so
to stop the old code loading. It will
I had/have exactly the same problem with my X100P / TDM400P dev setup.
To fix this error all I did was swap the PCI slots that the cards were
in.
And this error came back either due to a reboot or because I update to
the latest CVS. But again after much messing around with config's and
BIOS
Hello list,
please take a look at these units:
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596
are they locked? does anyone have one working with asterisk or SER?
Are these rebadged units from a different manufacturer?
anyone have any experience good
hi
if I'm on the phone to somewhere through this SMART IAD SIP/FXS
gateway, and I somehow lose contact with the SIP server (for instance
the SMART IAD reboots), then the channel will hang until the other part
hangs up.
is it possible to force a hangup on a channel in which the caller is no
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a reply in order to use sql database for all
sip clients from small offices asterisk box with nat
It says To enable this, you need to edit the Makefile in the channels directory of
your source tree and enable MYSQL_FRIENDS., but there is no MYSQL_FRIENDS in
channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
harry
hi
if I'm on the phone to somewhere through this SMART IAD SIP/FXS
gateway, and I somehow lose contact with the SIP server (for instance
the SMART IAD reboots), then the channel will hang until the other
part hangs up.
is it possible to force a hangup on a channel in which the caller is
no
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically loaded)
#
# Copyright (C) 1999, Mark Spencer
#
# Mark Spencer [EMAIL PROTECTED]
#
#
so, better is to look to another phone, than surcharge cisco ;-)
PJ
- Original Message -
From: AST 386sx
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Monday, October 11, 2004 2:00 AM
Subject: Re: Re[2]: cisco ip 7905 legal ..
If you are going to buy it new. It should
David J Carter [EMAIL PROTECTED] writes:
$1.64 to the £1 I think this morning so $35 stands.
But £55 x 1.64 is $90.2, not $35 ...
Regards,
Wolf
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello list,
please take a look at these units:
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596
are they locked? does anyone have one working with asterisk or SER?
Are these rebadged units from a different manufacturer?
anyone have any
On Fri, 8 Oct 2004, Michael Nolan wrote:
Hi !
I have checked my asterisk. It contains this patch or thBis patch is
already compiled into it. can you plz guide me as to how i can make use
of it ? I have pressed '#' but it doesnot give me any dial tone. Are there
any special changes that need
Hello all,
I having a lot of troubles to configure a simple voice menu.
In extensions.conf I have the following.
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,DigitTimeout,10
exten = s,4,ResponseTimeout,20
exten = s,5,Background(itranser/msg_bienvenida)
exten =
I having a lot of troubles to configure a simple voice menu.
In extensions.conf I have the following.
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,DigitTimeout,10
exten = s,4,ResponseTimeout,20
exten = s,5,Background(itranser/msg_bienvenida)
exten =
Hi,
I'am intergrating Asterisk with our CRM system. I have tryed this but
thats not working:
fwrite($socket, Context: local\r\n);
fwrite($socket, Setvar: Var\r\n);
fwrite($socket, Var: . $channel .\r\n);
Regards,
Paul
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Asterisk-Users mailing list
From few days ago there is no USE_MYSQL_FRIENDS in channels/Makefile. That is why I
am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List -
Thank you Christopher,
Imade the changes you told me, but, when I try to make an incoming call,
in the Asterisk console, I get
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/9'
-- Executing Dial("SIP/aurelio-92fe", "IAX2/501050:[EMAIL PROTECTED]/501050|60|r")
in new stack
-- Called
Is there anyway of monitoring an agent's status using the flash operator
panel ? I can monitor a queue easily but seem to hit a brick wall with the
agents.
Julian
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[EMAIL PROTECTED]
it's in there in -r v1-0, but replaced by some realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS in channels/Makefile.
That is why I am asking this.
-Original
At 13:53 11/10/2004, Roy Sigurd Karlsbakk wrote:
it's in there in -r v1-0, but replaced by some realtime stuff in
development CVS
I haven't found out more about that, though..
Old mysqlfriends is now remove from asterisk.
Now you have to use res_config_odbc for setup sip/iax friends.
you can read
please take a look at these units:
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596
The price of $30 after rebate certainly looks interesting.
are they locked?
If the firmware agrees with the manual at
http://www.voip2.net/Operator_Manual.pdf ,
it's not
Somebody seems start a mysql drivers for realtime external configuration
instead of ODBC.
You can speak to MySQL with ODBC.
bkw
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
Here is the Makefile from asterisk-1.0.0
--- Tomica Crnek [EMAIL PROTECTED] a écrit :
From few days ago there is no USE_MYSQL_FRIENDS in
channels/Makefile. That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
The Makefile isn't gonna help with cvs-head since the code was ripped out.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of harry gaillac
Sent: Monday, October 11, 2004 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial
Sorry
I have not look at CVS but I would like somebody help
me too about my problem.
help please
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit
:
it's in there in -r v1-0, but replaced by some
realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct
On Sat, 2004-10-09 at 02:16, deimios wrote:
On Sat, 9 Oct 2004 12:56:44 +0800, Walter Klomp [EMAIL PROTECTED] wrote:
Hi,
In the latest CVS I am trying to compile chan_h323, but it doesn't want to.
chan_h323.c: In function `oh323_call':
chan_h323.c:453: error: structure has no member
Use Asterisk v1-0 and please you're using chan_oh323 NOT chan_h323 they are
two totally different channel drivers.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Pablo Endres
Sent: Monday, October 11, 2004 7:49 AM
To: deimios;
Hello,
in advance I'd like to apologize myself for probably stupid questions
which follow, I'm just a newbie to Asterisk:
I'd like to use Asterisk as VoIP gateway between two PBXen. Ie:
Phone Net 1
|
PBX 1 --- TelCo
|
Asterisk 1
|
[VoIP]
|
Asterisk 2
|
PBX 2 --- Telco
Just thought I would let the
list know, as we got our pre release versions today of the new Zoom X5 that
supports VoIP. The device comes with an RJ11 phone socket on the back and lets
you configure your ADSL router to become a SIP phone (using your existing PSTN
phone). Better still, it
res_config_odbc and ast_data is the new way
the old way is still in 1.0.1 and CVS -r v1-0
ast_data is available at
http://svn.asteriskdocs.org/res_data/
roy
On Oct 11, 2004, at 14:47, harry gaillac wrote:
Sorry
I have not look at CVS but I would like somebody help
me too about my problem.
help
Can someone post or forward me the relevant sections of their nufone
configs?
I seem to be brainfarting on making it work. All my outbound attempts
end up with results like this:
bebop*CLI iax2 debug
IAX2 Debugging Enabled
bebop*CLI set verbose 9
Verbosity was 0 and is now 9
-- Executing
Hello Pavel,
well .. any GOOD propisition for the same or lower price would be nice
IP300 and ip500 are more expensif than this one
--
Best regards,
Dannymailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division - internet solutions
place princesse
Yes but it's will be better to have mysql driver
At 14:20 11/10/2004, you wrote:
Somebody seems start a mysql drivers for realtime external configuration
instead of ODBC.
You can speak to MySQL with ODBC.
bkw
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[EMAIL
Hi all,
Just two questions:
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
Anybody could answer to my first question ?
Harry
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit
:
res_config_odbc and ast_data is the new way
the old way is still in 1.0.1 and CVS -r v1-0
caller id on/off, ...
^
Should I interpret it that simple ISDN cards supported by I4L doesn't
support CLI/CLIP/CLIR?
No, it yust says that you cannot select by software if to transmit
caller id. If the line is configured to generayyl transmit ID it
should be ok for you.
Elmar
look at unixODBC or iodbc for more information
Also the reason (i guess) why they move to ODBC is that's ODBC have many
connector to most SQL database.
At 15:26 11/10/2004, you wrote:
Hi all,
Just two questions:
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
Anybody could answer
You must be one of those people that doesn't know much about ODBC and is
under the impression it's SLOW!
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Arnaud Pignard
Sent: Monday, October 11, 2004 8:23 AM
To: Asterisk Users
Hi all,
hi
Just two questions:
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
To bypass licencing issues in MySQL?
Anybody could answer to my first question ?
To bypass licencing issues in MySQL?
roy
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[EMAIL
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
1. It's not Microsoft at all.
2. It's unixODBC (I don't see Microsoft here at all)
3. Wider database support without having to know each database type.
4. It's not much slower than native DB drivers. (15-33% slower)
But In
look at unixODBC or iodbc for more information
Also the reason (i guess) why they move to ODBC is that's ODBC have many
connector to most SQL database.
Bingo
bkw
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[EMAIL PROTECTED]
HI Everybody!
My company is seeking to replace its legacy PBX by a VoIP solution; since we
prioritize the Open Source Paltform we have found Asterisk doing our own
research and we are very interested in it.
Knowing that we are decided to make the move to VoIP, can somebody tells me
the
[EMAIL PROTECTED] wrote:
But In my tests you would never see this unless you're doing 10k
selects and 5k inserts and that's on a 1ghz box.
Per seconds? Per day?
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
t: +31 (0)10 2245544f: +31
Here is my first question.
Two smalls offices with sip clients
+ Asterisk, one offices with Asterisk and mysql
database.
I would like to define all sip peers in mysql database
so Asterisk from small office could read sip peers
configuration from database office.
May I use autocreatepeer in
Somehow you are out of sync with CVS. app_realtime is not in the 1.0 branch.
ast_load_realtime is defined in config.c so somehow you got the soruce to
app_realtime but didn't get an updated config.c and many others.
If everything is working now, just make a noload=app_realtime.so
Matthew
-
Per second.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andreas Sikkema
Sent: Monday, October 11, 2004 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL
All db specific code has been removed from the code in favor of the
currently-in-development RealTime method of configuration from database.
You are most likely not using the 1.0 stable branch.
You need to use the new RealTime configuration method. And currently, there
is only support for odbc. I
Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 onRedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.
Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following
I am in the final stages of writing res_config_mysql.so So far, all of my
internal testing with it works. Stand by..
Matthew
- Original Message -
From: Arnaud Pignard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, October
Cheap shot.
Digium does Asterisk FOR FREE.
No. As with most of us who support free software projects, we support
them because it suits our business goals. We don't do it for free. The
investment in time, effort, and resources is paid back, frequently in a
way which can't directly be
No , i use unixODBC on several application/servers.
but as you said :
4. It's not much slower than native DB drivers. (15-33% slower)
I have never done any bench about it. So i can't make any argumentation on
it and seems you have done some bench.
However add unixODBC on the middle won't be
I don't think you want a latching relay, unless you know how to build
the support circuit -- a latching relay has two coils and requires a
short pulse of power on either coil to change state. The advantage is
that it doesn't need any power to hold state, but of course the circuit
isn't
Doing some further searching it looks as though as Steve pointed out
earlier the TDM400P may work for this. Has anyone else used the TDM400P
to handle analog DID trunks?
Steve Underwood said:
Hi,
Technically you can do it, but whether you can get that as a service
depends on where you live.
Thank you for your reply.
I forgot to mention ... Asterisk dies with that error message ...
Everything goes ok with download/compile but when I want to run
Asterisk it dies.
Message: 7
Date: Sun, 10 Oct 2004 21:14:53 -0500
From: Brian West [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] newbie
Hi List,
I've successfully got Asterisk up and running, can make out going calls
fine, It can also register FWD OK, but when a Call comes in from outside it
is rejected with this message.
Oct 11 09:09:40 NOTICE[98310]: chan_sip.c:7175 handle_request: Failed to
authenticate user
Hi I cant get the Message waiting indicator to light
on my 7910 phones. What am I missing? Here is a snip of my skinny.conf
[Guest]
device=SEP00044DE12922
version=PC040300
host=192.168.254.18
nat=0
callerid=Henry
Devito 1277
mailbox=1277
callwaiting=1
transfer=1
Wolf N. Paul wrote:
Except that £55 is more like $75-80 and not $35.
Regards, Wolf
Reminds me of a wonderful anecdote about a college english professor
who, upon reading in one of his student's compositions that a character
had fallen down stairs and laid prostrate on the floor, that the
Greg,
Which kernel are you using? I have two machines at home and the zaptel
kernel module only runs properly on one of them...
The P-3 box worked...
kernel-2.4.20-30.9.i686.rpm
The Athlon did not...
kernel-2.4.20-31.9.athlon.rpm
Both machines were updated on the same day (apt-get) and for the
I am getting some weird behavior and a rash of interesting messages in
the log files. If anyone has some ideas, it would be appreciated.
Using Asterisk v1.0.1 on Suse Enterprise Linux v8.0. HP DL380 Server.
4GB Ram - Dual 3.2ghz processors.
This first entry is when asterisk simply goes
Why don't you take this off-line were it belongs
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco
Sent: Monday, October 11, 2004 9:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards
Cheap shot.
Why don't you take this off-line were it belongs
You don't think discussions about the Asterisk user community belong on
asterisk-users?
It belongs right here. Participants who want to alienate potential new
users just because they didn't buy a Digium product have a negative
effect on the
Beau Walker a écrit :
[...]
here is my SIP.conf
register = 499yyy:[EMAIL PROTECTED]/499yyy
[fwd] ; inbound connections from FWD
type=user
nat=yes
host=dynamic
context=fwd-inbound
canreinvite=no
qualify=yes
insecure=yes
This is not needed. You have type=user and below type=friend (which
include
Donny Kavanagh said:
Could these files be cached as well?
Not sure what files you're refering to but the AGI Perl script isn't being
cached as I've been able to change it and call the extension to see the
changes without a reload. No res_perl going on here unless it magically
part of the stock
Hi all,
We've completed asterisk 1.0.0 and patched it to work with MGCP 1.0 and
NCS 1.0, also we've registered the DPX 2203 Cable Modem with embedded MTA
and it works fine except:
- It can't detect off-hook state until I press flash in phone and,
- It wont ring when I dial from another phone
hello,
I wrote to [EMAIL PROTECTED] in order to
someone help me without reply ?
May be you could help me
Here is my problem.Two smalls offices with sip clients
+ Asterisk, one offices with Asterisk and mysql
database.
I would like to define all sip peers in mysql database
so
--- ismaelg [EMAIL PROTECTED] wrote:
Hello all,
I having a lot of troubles to configure a simple
voice menu.
In extensions.conf I have the following.
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,DigitTimeout,10
exten = s,4,ResponseTimeout,20
exten =
I have my asterisk voicemail set up to e-mail me .wav attachments
(in
the
wav49 format), and I receive the messages fine, but the volume is so
low
that I have to turn my speakers as high as they will go in order to
hear
it
(which makes it interesting if I forget to turn them down
Asterisk Users;
Just wanted to let you know I fixed my problem.
To follow up on my own testing of the situation, I find that the
continued ringing after pickup only occured on the SNOM phones in the
group. The Grandstream phones stop ringing when another phone picks up.
Having turned
What is anyone out there using that's small, quiet and robust for a
SOHO system with two X100P and a TDM400? I'd love to see some recos
for easy to find hardware to build asterisk office pbx.
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[EMAIL PROTECTED]
is there any way to read global vars like ${EXTEN}, ${GROUPCOUNT} from an
AGI?
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Someone pointed me here
http://www.snom.com/downloads/share (had to guess at URL as the Snom
site appears to be down or uber slow but if that's not it its damn close
:-P )
Which lists all versions of firmware for all their phones. Handy if you
have a specific version in mind but don't know the
You have obviously never posted to any kind of mailing list before.
Sometimes you may have to wait a few days for someone to answer you.
Sometimes people just don't know. Griping to the owners of the list about
the people who take time out of their day to give you FREE support isn't
going to make
How do you install this? I downloaded it from sourceforge,
but I can not find a documentation or how-to
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It says FXS. How are you setting your
switches? All the rear panel switches are set to normal but Im unsure of
the front.
Thanks,
Mason Herring
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim McKee
Sent: Sunday, October 10, 2004
8:35 PM
To: 'Asterisk
Henry Devito ([EMAIL PROTECTED]) wrote:
How do you install this? I downloaded it from sourceforge, but I can not
find a documentation or how-to
currently i am writing one ...
--jan
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Dear Sirs,
The Asterisk bounty has been updated accordingly.
Some info about our environment:
Our Asterisk server is logically connected to a Veraz NGN platform
through SIP and we are facing two major problems for calls from/to
Veraz;
When calling from Veraz to any SIP extension, no ringback is
Thanks, I will be patient and wait.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jan Czmok
Sent: Monday, October 11, 2004 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] chan-sccp2
Henry Devito
Has anybody had any experience connecting the t100p to
a verizon smart jack.
I've been told the t100p uses an RJ48 but not the revision
(i.e. C, S, X )
I've created wires (RJ48C x-over) but no green light on the t100p
1-4
2-5
4-1
5-2
i've created wire (RJ48S) no green light
(only because the
is it possible to windows messenger clients of an asterisk server to chat
(text chat) with each other?
what about the status presence? is it possible to each windows messenger
client of an asterisk server to see the presence on other clients?
if not, what is missing in asterisk?
Asterisk doesn't support MSN9 the protocol Windows Messenger (and MSN
Messenger) uses to communicate with a messenger server such as MSN or
Windows 2003 running the Live Conferencing server.
It should be possible to write an MSN9 server independently of Asterisk
since the information needed by
Hi, I'm try to get asterisk up and runing on my linux pc, but I can't download the file (asterisk,zaptel libpri), i got connect to your ftp server but I can't download the files from asterisk or diguim, i login as anonymous, i saw the pub file but i can't got it,if somebody give a hand to
Is there a fix/patch that can be applied to allow the voicemails to be
recorded LOUDER? I would like to go live with my Asterisk system, but
this is a major problem.
Its not asterisk that's the problem I suspect. If you get low recordings
you need to look into using app_test to help find
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