Nick Bachmann wrote:
Yes, I've played with it a bit. It's pretty simplistic... the clustering
just keeps several servers in sync with each other. I suppose that would
be easy to do with Asterisk, especially if configuration data was stored
in a RDBMS that could do replication. Even now, setting
Doug Shubert wrote:
I would set the "Enterprise Class" bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
In our network, Linux is approaching
"Enterprise Cla
Dawid Mielnik wrote:
Hi all,
I have my asterisk setup as following:
IP 2 x E1
x-lite <---> Asterisk ---> PSTN
When I place a call from x-lite to PSTN, the quality of the sound in the
direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
he
JR Richardson wrote:
Piping in 2 cents,
This is a great example of the Internet, Fast Food generation, showing their
appreciation for all the magic that happens in the labs, hearts and minds of
the courageous, hard working, dedicated and motivated group of people truly
interested and guided to ac
Peer Oliver schmidt wrote:
Hi,
we have implemented a first version of call support from a web based
system for Asterisk (via the manager interface) and other, callto: and
tel:, providers.
Now I am looking at the other way around. If a call comes in, I want
our web based system to automaticall
Greg Boehnlein wrote:
On Tue, 30 Dec 2003, Tilghman Lesher wrote:
On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote:
What happens when you change the configuration of the GS phone to
send DTMF via SIP INFO?
I had that set originally. I get the same behavior no matter wether I u
Senad Jordanovic wrote:
WipeOut wrote:
Hi all,
Let me be the first to wish everyone, especially the Digium team, an
awesome year in 2004..
Later..
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Hi all,
Let me be the first to wish everyone, especially the Digium team, an
awesome year in 2004..
Later..
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Greg Boehnlein wrote:
Hello,
I have been retained by a Building Management Company to install a
combined Voice/Data solution for a Tennated Office Space. This space will
rent offices, with telephone and internet service to inviduals or small
groups of individuals. As fate would have it, the se
Justin Sinclair wrote:
From: "David Luyens" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] fedora core 1 install problem
Date: Mon, 22 Dec 2003 14:20:55 +0100
Reply-To: [EMAIL PROTECTED]
Hi Ernest,
I have installed as you described, and now it worked.
Seems that instal
Brian West wrote:
cdr_odbc is for logging CDR data to a database. Its pretty much blind to
the type of database you choose as long as it has an ODBC driver.
We had it speaking to an AS/400 running DB2... we also have it working
with MSSQL (not my goal but hey it works), mysql, pgsql and flatfiles
Ing. Angel Gomez Garcia wrote:
Hi all.
I have this configuration:
Telco <-(E1)->TE410P//Dual Xeon Server
2.4Ghz<-(Ethernet)->Switch<->GS//BT
The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp
and we are having the following 2 issues:
1.- When mak
FRANCISCO PEREZ-LANDAETA wrote:
Hi Friends,
I am new to linux and new to asterisk. I need some help setting up
asterisk in my linux box. Does anyone have a step by step guide ? On
my PC i have installed a phonejack (from Quicknet) as well.
Your help is appreciated.. I kind lost..
thanks,
My
bam wrote:
The phone powers up and I can make calls through my Asterisk gateway
to other endpoints. However the four leds under the keypad are
permanently illuminated and the backlight slowly flashes on and off.
When I pick up the handset there is a repeated tone before I get a
dial tone.
I k
Brian West wrote:
Well you certainly could. And you'd then have to add the cost of the
ATA to your "cost per seat," at least doubling the $65 figure--tripling
it if you meant a Cisco ATA.
NOT, Cisco ATA's can be had fro 99-120 if you are lucky. Then you can
also get a cisco 7905 which can b
I just loaded the b13p4.30.zip firmware and now I am not able to log
into the GS admin interface.. anyone else having this problem?
Going to try the next older version..
Later..
mikeu wrote:
http://www.grandstream.com/TEMP/FIRMWARE/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
vocalvoip wrote:
hi guys
just got a question, im using grandstream phones with canreinvite=no or woteva, all nat etc is working perfectly. but i believe because of the canreinvite, when a call has taken place the voice will be proxied via the sip server to the 2 parties involved. ( which means th
Mark Spencer wrote:
Amsterdam!!
I had my laptop and suitcase stolen in Amsterdam the one time I went
there, after hearing someone talk about how safe a city it was over
dinner. Most importantly, also stolen was my (apparently irreplacable)
copyleft shirt (yellow/gold with large blue backward
Make sure and let us know anytime you are stopping by London.. :)
(Just not between the 2nd-22nd December cos I will be away)
Later..
Mark Spencer wrote:
I'll be there until jan 5. The 19th would definitely be too early, maybe
the 20-22? Possibly even after the new year, jan 2 or 3.
Mark
On Su
David M. Wilson wrote:
Hi there!
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my Googling whether my configur
Anton Yurchenko wrote:
Hello,
is there a way to disable call waiting in sip? I`m using grandstream
101 and even when the phone is in use I hear ringing in the headset.
It is pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
Try
Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=586
woop... Anyone wish to test and or make this better?
(I know some of the code can be put into functions)
bkw
Do you think this will be merged into the CVS (seeing as its based on
LGPL) or will it be an addon?
Later..
Girish Gopinath wrote:
Hi,
I have received some replies for my previous mail (* configuration),
asking for my goals in configuring Asterisk. So here they are:
We are planning to host an Inter continental virtual PBX service that
will enable our users to register for an account which give them
Asterisk wrote:
Hello Asterisk-ers,
Thanks to WipeOut! you've kinda answered something I wondered about.
I've been looking for a post like yours for the last 3 hours (I didn't
want to get told off for not looking first)!
I don't have enough Asterisk boxes (yet) to test this
costas wrote:
I was just looking at AGI with PHP myself. I just have a real dumb question. How does Linux know to send $stdout(or echo) to *? What if there are other apps open as well waiting for input. WOn't they get the output?
Also, how does the AGI know to read from $stdin is * input?
Costas
A note to all those who are avoiding writing up an AGI becasue it looks
two complicated..
I have just written up my first and its awesome.. It makes Asterisk open
to all sorts of possibilities.. let your imagination run wild..
I put off writing an AGI script because a) I could not find any docs
Pavel Litvinenko wrote:
Brian West wrote:
asterisk*CLI> load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR
Backend)
== Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
Walker Haddock wrote:
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] > direct to snom200
or
si
Franz S wrote:
Hi guruz,
I have requirements from a company, which is going to deploy call
center nationwide using asterisk and the new 4 port cards. However
before going to purchase the hardware they want if following is
possible in the asterisk software
1) they want to whisper with o
I am trying to use the SAY NUMBER command from an AGI script but it does
not seem to be working..
If I use "EXEC SayNumber 2" and execute the asterisk command from the
AGI it works and I hear the 2 said on the phone..
If I use "SAY NUMBER 2" I see "-- Playing 'digits/2' (language 'en')" on
the
Anton L. Kapela wrote:
Jeremy McNamara said:
Do you really want all those spans going down cause someone tripped
over
a power cable or your hard drive nukes itself?
How's this worse than an as5300? I could install ata-flash and get
high-ish end pc hardware (rcc serverworks boards, etc). He
Daniel ANDRE wrote:
Hello all,
I wonder to have some feedback on using ISDN BRI Cards with Asterisk
and the Echo problem.
I have tried a simple BRI card with i4l driver and encounter huge echo
problem. I have tried to solve it with a Sw chocanceller without
success. What I'd like to know is w
I have the following setup..
[extensions]
; all extensions defined here.
exten => 1234,
exten => 1235,
[dial-out]
; PSTN dialout config
ignorepat = 9
exten => _9,
exten => h,
[local]
; phone context in sip.conf is here..
include => extensions
include => dialout
The question is w
Mark Spencer wrote:
Amen! While -dev and -users may be a little too sparse, perhaps adding a
-business list would be beneficial for discussing those types of issues.
However business-related issues are not so common at this point, so perhaps
a list devoted to NONTECHNICAL discussion (-nontech?) w
Hans-Henrik Andresen wrote:
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
HMM. greate sarcasm.
I had read about a driver for asterisk for voicemodems, that why
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
No, it can't be done..
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It seems to me that ITSP's like to use a US dialing code eg 1-xxx
Wouldn't it be cool to have an Internet dialing code??
I don't know what the structures are or how the allocations work but it
would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx
was an internet phone.. That wa
Klaus-Peter Junghanns wrote:
Hi,
if anyone is looking for a really small * installation take a SuSE
rescue system, add sshd, zaptel and *. Gzip it and the image is only
28MB which fits into a 64MB ramdisk and leaves some MBs for temporary
logs, CDRs and voicemail.
regards
kapejod
I know this
I have two possible senarios for making a call from an AGI..
Senario1 - Using GoTo
In the extensions.conf I have..
[dial-out]
exten => _9.,1,AGI(myagi)
exten => _9.,2,Dial(SIP/blah/${EXTEN:1})
In the AGI I have..
EXEC GoTo dial-out|9555678|2
So using this method I don't have to really edit the AGI
Hi,
If anyone is looking for a small Asterisk installation I have managed to
get it down to 296MB (If you remove the kernel source code.. could
probably be made smaller if some of the devel packages and asterisk
source is removed as well.)
To do it I used Trustix Secure Linux 2.0 (http://www.t
[EMAIL PROTECTED] wrote:
Hi everybody,
I am getting into the * world and I need some help...
I woudl like to know the possibilities of doing VOIP over the Internet
between 2 or more (point to multipoint) locations
I have read documentation from * website, but the examples I came
across are usual
Surajee Ratnayake wrote:
Hi,
Do Digium have any plans to release a 4 port fxo card.
If yes, when?
I think they are in the pipeline.. Initial speculation was that they
would be out in September but I guess there have been problems..
I guess the best answer is they will come out when they co
Larry Black wrote:
[hardware]
type=friend
callerid="Hardware Phone" <5>
secret=phone
echocancel=yes
host=dynamic
dtmfmode=rfc2833
context=sip
My standard config for GS phones on the same LAN as the Asterisk server is..
[hardware]
type = friend
callerid = Hardware Phone <5>
secret = phone
host
Sathya Weerasooriya wrote:
Hello,
I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
I cant figure out where to install the asterisk-addons. Is it in /usr/src or
/usr/src/asterisk ?
Once I
[EMAIL PROTECTED] wrote:
Does anyone have a setup (or am I missing a simple thing here) to
transfer a caller directly to someone voicemail? Example: I receive a
call, the caller wants to speak with x, who I know is not in the office.
Other than transferring them to x's extension, which rings to t
Philipp von Klitzing wrote:
Hi there,
I still have issues with the IAX connection between two servers (one
static (server A), one dynamic (server B), none behind NAT):
B registers with A, and "iax2 show registry" shows that everything is
fine. However, after a while if I check on server A with
[EMAIL PROTECTED] wrote:
Hi guys,
Having a bit of a problem trying to get conference bridges working. In my
meetme.conf file I have the following line
[rooms]
conf => 6000
Make sure you have at least one blank line at the bottom of your
meetme.conf..
Later..
___
John Brown (CV) wrote:
Any thoughts on how to make HOLD and TRANSFERs work
with a SNOM 200 and Asterisk ??
Thanks
To hold simply press the function button that has that call on it (the
button on the right with the light on)..
To blind transfer press the "Xfer" button (the first one on the le
John Brown (CV) wrote:
Hi list,
how does one get a SNOM 200 MWI to work with * ??
When I press the MWI button it doesn't connect with
voice mail on my * box.
thanks
You need to create an extension called "asterisk" and redirect it to
your VoiceMailMain extension..
eg. If your voicemailmain
I have been playing around a lot with the CDR today and I may have
stumbled across a very serious problem, specifically where there is
billing taking place..
If a call is placed between 2 phones and the network connection is
broken from both the phones with out hanging up first the call is neve
I need to get your thoughts on something.. :)
I am trying to create a system to process the CDR call logs for
department accounting..
I think there are two ways of doing it.. Either I can create an AGI that
will run on the "h" extension and will lookup the last entry that
matches the account c
Using the verbose comand in an AGI you can feed information back to the
console in the same way that applications do.. so by specifying
different information to be returned at various verbosity levels the AGI
will produce results similar to those of Asterisk applications..
Is there any standard
David Winkler wrote:
Hello. Was just curious to know if anyone is working on a graphical
interface to Asterisk using X windows, or something else similar.
Thanks!
David
There are a few projects on the grow.. mostly web based.. Nothing
complete yet..
__
Question 1..
Do the "say number" and "say digits" commands in AGI scritps work?
If I use "EXEC SayNumber 123" it works but is I try "say number 123" it
doesn't.. I think I have the syntax right becaasue thats how its shown
when typing "show agi" on a console and also on the agi pages I have
lo
Chris Albertson wrote:
Testing a querry by doing 2000 identical querries and then
deviding the total by 2000 is not a valid way to
measure the time to do one querry. The result will appear
to be as much as 100X or even more to fast.
The reason is:
1) Operating system will have cached the exact d
David Gomillion wrote:
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
Query2 ("code" field not indexed) = 109.321s
Query2 ("code" field indexed) = 2.302s
I disagree with your disagreement :P We have to keep in mind the b
Roy Sigurd Karlsbakk wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) =
Andy Powell wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
I read on a site yesterday (wish I had saved it now.) that said that
MySQL were re-visiting their new licence policy to make it possible for
projects to use MySQL again..
Has anyone else seen this?
This looks like good news, it means that the MySQL stuff may be able to
be merged back into the
Max Lock wrote:
Hi Folks,
Bit of a newbee here, so please be gentle. :)
I'm trying to get the message waiting indication working on a
budgetone-101. Is it as simple as putting `mailbox=n' where n is the
mailbox number into sip.conf?
Is there anything else I should check or set?
Max,
If yo
Rich Adamson wrote:
Calling from a SIP or ZAP phone out of an Asterisk box using an X100P
alway shows "NO ANSWER" in both the MySQL CDR and the Master.csv file
and the billsec is always 0..
The process, I dial my cell phone, answer the call, wait a few seconds
and then hangup the SIP or ZAP ph
Robert Mann wrote:
You are correct.
Confirmed in: CVS-11/08/03-10:20:40
Robert
Thanks I will add it as a bug..
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Hi,
Can someone test this to confirm a bug..
Calling from a SIP or ZAP phone out of an Asterisk box using an X100P
alway shows "NO ANSWER" in both the MySQL CDR and the Master.csv file
and the billsec is always 0..
The process, I dial my cell phone, answer the call, wait a few seconds
and the
Hi,
Is the AGI command "set autohangup" the same and the "AbsoluteTimeout"
command?
Which is best to terminate a call after a specified length of time?
Thanks..
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Florian Overkamp wrote:
Hi,
At 13:48 11-11-2003 +0800, you wrote:
i made a lot of silly mistakes along the way which could have been
avoided if only there were some kinda howto or samples. at the risk
of looking stupid, i decided to shared my experience in hopes that
it might help some newbie g
Thanks for that ..
I have also just started writing a AGI in PHP and have also discovered
some of the issues.. you have shown me a few more that I probably would
have bumped into.. Should save me a little stress.. :)
Later..
hkirrc.patrick wrote:
i've just spent the pass 2 days trying to get
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
Query2 ("code" fiel
Thorsten Lockert wrote:
SELECT *, LENGTH(code)
FROM a
WHERE code = left('00442085673456', LENGTH(code))
ORDER BY LENGTH(code) DESC
LIMIT 1;
Awesome, thanks Thorsten..
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Reinhard Max wrote:
On Mon, 10 Nov 2003 at 13:34, WipeOut wrote:
I guess I am going to have to look through the table multiple times
dropping the last digit on each select until I get a result..
You could also try this one to see which one is faster:
SELECT *, length(code)
FROM a
[EMAIL PROTECTED] wrote:
Is anyone running Asterisk under Fedora Core 1 (http://fedora.redhat.com/)?
If so, did everything with Asterisk work properly? I'm looking to migrate
from Red Hat 8.0 to Fedora this week.
Thanks.
From what I have seen in playing with it for the last day and a bit is
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is
provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and
ztrtc can't run on smp systems.
So if you only have smp systems with ohci
Reinhard Max wrote:
On Mon, 10 Nov 2003 at 12:33, Reinhard Max wrote:
SELECT DISTINCT *,length(code)
FROM a
WHERE '00442085673456' LIKE (code || '%')
ORDER length(code) DESC;
^
BY
Oops, that one got lost when I re-formatted the query.
cu
Reinhard
Hi Reinhard,
Th
SQL help needed..
If I have a MySQL table with dialing codes and a corresponding
description (see below) and I want to lookup the best match for a phone
number.. What would the SQL look like to do it? or would it take more
than just SQL to get to the best result?
Thanks..
Later..
Example num
costas wrote:
2) Put on hold and pick up on a different phone set.
The right thing for this is call parking but it doesn't work to well
with IP Phones..
Could you clarify what doesn't work well? Is there a SIP deficiency?
It does not work so well becasue the IP Phone (GS specifi
costas wrote:
Hi,
Does anyone have sample * configuration on how I can get an incoming call to ring all SIP phones (small setup, say 4 phones) at the same time.
1) I would like to pickup up any phone and the ringing should stop (of course)
Example..
exten => s,1,Dial(sip/123&sip/234&zap/2)
Will
Rich Adamson wrote:
Is any one else having problems with the Snom 200 MWI??
If flashes and shows me there is a message then I go and listen to the
message but the MWI does not clear.. The only way I have found to clear
the MWI is to reboot the phone..
Gus,
Works correct for me. Running v2
Is any one else having problems with the Snom 200 MWI??
If flashes and shows me there is a message then I go and listen to the
message but the MWI does not clear.. The only way I have found to clear
the MWI is to reboot the phone..
___
Asterisk-User
Can I add to this and say that another thing that could be hindering the
takup is "Single System" VoIP scalability and a certain amout of
"Enterprise" flexibility..
Let me explain those two..
Before you start reading these and thinking "This guy is mad!!" let me
just say that I love Asterisk a
Rich Adamson wrote:
Brian,
My Snom 200 (about 2 months old) is running v2.02q from a couple of days ago,
and the speakerphone is fine. Have not tried a headset with it though.
Just ran a short test from a 7960 to the 200, both on the same wire, to
validate the quality.
Looks like Snom have
Steven Critchfield wrote:
On Fri, 2003-11-07 at 13:55, WipeOut wrote:
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using "EXEC Dial
channel/number" the problem with this is that the script then completes
and does not wait for the c
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using "EXEC Dial
channel/number" the problem with this is that the script then completes
and does not wait for the call to end..
Is there an alternate way to dial the call and then when the call is
co
C M wrote:
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
Take a look in the root of the source (/usr/src/asterisk) at the
README.cdr and in the /cdr/cdr_csv.c file for descriptions of the fields..
please help me. i want to store these into mySQL
data
Mark Evans wrote:
Hi All
I have a snom 200 phone here which works perfectly when using the
handset to playback the voicemail messages etc.
However when I play back the voice using the speakerphone it sounds
choppy. Anyone had this problem before?
Regards
Mark
Yes, I think its the echo cancel
Is anyone having sound quality issues with the GS phones when calling
out to the PSTN through a X100P?
Basically I am doing this..
I call out to my cell phone and then with the cell phone to mu ear I
gently blow a constant stream of iar into the mic on the GS phone.. What
I should hear in my e
mattf wrote:
Hello,
I am trying to figure out the maximum capacity of concurrent SIP users(SIP
to Zap T1 channels) on my asterisk system. I assume that the processor/RAM
is the weakest link in the chain in terms of capacity right now.
* see my configuration below
Right now with 18 concurrent use
Hi,
Has somthing radical happened to the way sip traffic is handled??
I was just doing some testing and where G.711 used to use about 84kb/s
of bandwith its now using between 150kb/s and 190kb/s.. The lowest I
could get it to in the Snom was 160kb/s and 75 packets per second which
was by setti
Tilghman Lesher wrote:
On Thursday 06 November 2003 05:16, WipeOut wrote:
It would appear that the "uniqueid" field is not being populated in
the MySQL CDR DB.. Is this an obsolete field or is a bug?
Use the source, Luke. You need to define MYSQL_LOGUNIQUEID at compile
time for
Nick Knight wrote:
Voipvoice handsets we tried - and are now sat on a shelve gathering
dust. The main problem was the quality of the audio - to quiet and poor
- not telephony grade for the office - perhaps good enough for home use.
Just my two pennys! But still looking for a usb handset!
Nick
Shoval Tom wrote:
Guys, it still not working.
Go here
http://www.checkdns.net/quickcheck.aspx?domain=voip-info.org&detailed=1
And see that it returns errors.
PLEASE help.
None of the reported errors are critical.. They are just saying that
only one DNS server is active..
Try setting you PC
Andrew Kohlsmith wrote:
Anyone got any pointers on where to find USB handsets or headsets that
can be used as the audio device on a softphone?
I am a fan of the DSP-100 from Plantronics (mono USB) but unfortunately I
have been unable to get it to work nicely under Linux. Works _great_ under
Roy Sigurd Karlsbakk wrote:
I have a couple of USB handsets from Clarisys (SP-phone), but I have no
idea of how to interface them...
Aren't the Clarisys products a little pricey??
On Thu, 2003-11-06 at 14:55, Dan wrote:
Hi,
...
Hi Dan,
I had looked at that option but it seems that the o
Dan wrote:
Hi,
Try something better.
Use a Bluetooth Headset and an USB Bluetooth dongle with AudioGateway
Profile support.
Then you will have a cordless headset which can be used 10m around the PC.
I have the following combination wich works very nice with my DIAX phone.
- headset Plantronics M1
Anyone got any pointers on where to find USB handsets or headsets that
can be used as the audio device on a softphone?
Later..
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Kerker Staffan wrote:
Hi
Is it possible (or recommended) to run both Asterisk and
say SER on the same physical machine? How about port conflicts?
Maybe the easiest way is to change the default SIP port on Asterisk?
But how will that work if I register some SIP accounts directly
from asterisk (lik
Gavin Hamill wrote:
On Thu, 2003-11-06 at 11:16, WipeOut wrote:
It would appear that the "uniqueid" field is not being populated in the
MySQL CDR DB.. Is this an obsolete field or is a bug?
I have never looked at this package so I've never read any docs for it,
buy my gu
It would appear that the "uniqueid" field is not being populated in the
MySQL CDR DB.. Is this an obsolete field or is a bug?
Later..
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Shoval Tom wrote:
Setting it in hosts doesn't do me any good.
Trying to surf to http:// 64.65.102.50 gets me to apache test page.
Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
Get a 404 page doesn't exist.
Its most likely on a name based virtual server.. edit your hosts f
Steve Underwood wrote:
96% uptime would mean nearly 4 hours per month down. I have never
experiemced anything that bad using the nastiest crappiest no-name
server parts. unless you want to make a point, like some authors
do. Then you say the hard disk failed and it took a week to get and
i
Steven Critchfield wrote:
I think the number you cited needs qualification to be accurate. Because
if it where accurate as it stands, I'm due for major downtime in my rack
as I have several systems approaching 2 years uptime without a single
hardware failure. These machines also where not new when
Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of is 'Relatively few installations
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