Hi,
Is anyone aware of an IAX client that's made for the Windows CE/Pocket PC
platform? Or even the Palm platform for that matter.
Thanks.
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Hi,
I'd like to know which version of Asterisk performs best and most stable
with TE110P.
I don't need any other features (it'll just terminate interasterisk calls
without any other feature - so there is no need for CVS Head features or
? ).
Any info on setting up secure interasterisk IAX conn
Matthew Boehm wrote:
Is there anything I can do to track down the problem? e.g., is there a
command in *CLI to read the database record? push a record, see the
differences, ...
You can try this:
realtime update sippeers allow g729 name 621
That should be the SQL equivalent to "UPDATE TABLE sip
Whenever I try to install [EMAIL PROTECTED], I get this error at about 43%
There was an error installing rpmdb-redhat-3.4-0.20050105. This
can indicate media failure, lack of disk space, and/or hardware
problems. This is a fatal error and your install will be aborted.
Please verify your media a
hello all,
i am happy to be part of the asterisk community. i
have sucessfully configured my asteisk server with the
following (fxo card, fwd-ipkall, fwdout,) everything
works great except when i attempt to check my
voicemail via any iax softphones for windows (diax,
iax phone). this problem does
I am in China , I cann't buy digium card.
I want to resales asterisk in China for chinese enterprise.
who can give a card for test ? I only hope COD.
I hope buy a TDM400 and a FXO .
Thank u.
Best Regards
Zhao Zigang 赵子刚
Alcatel Shanghai Bell Co., LTD
*:388,NingQiao Rd.,Shanghai 201206
*:086-
Hello, users!
It is simple.
I want to make a call among asterisk and netmeeting(192.168.1.107).
chan_h323 & chan_vpb are loaded.
The call from netmeeting to asterisk(chan_vpb) is established a connection
and can communicate each other.
The call from asterisk(chan_vpb) to netmeeting is also connecte
hello there,
I have searched lists about an application chan_spy, people talked about
it on lists that we can use it to monitor sip to sip calls. but I am
unable to find any clue of it.
can some one please tell me from where I can get this chan-spy application
thank you
regards,
--
Atif
Beware of 1.0.5.23 Grandstream firmware. When I installed it, SIP
registration stopped altogether. Going back to .22 fixed things again.
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check for interrupt conflicts, cat /proc/interrupts
- Original Message -
From: "Anton Krall" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, March 15, 2005 8:26 PM
Subject: [Asterisk-Users] Voice getting cutoff
Guys.. I just noticed
What settings are you using to burn the iso? If you are using Nero or
several others you have to tell it to burn the disk at once, not track at
once. I had the same problem
- Original Message -
From: "John Novack" <[EMAIL PROTECTED]>
To: "Scheda" <[EMAIL PROTECTED]>; "Asterisk Use
I'm running 1.0.5.22 (beta), and it is the best version I've found to date.
I notice .23 is also available.
http://gs-firmware.gratissip.dk/
- Original Message -
From: "el Flynn" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, March 16, 2005 3:20 PM
Subject: [Asterisk-Users] Grandstream and Tra
Hi all,
Just wondering if anyone's come across this issue, and what might be a fix for
it:
We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The
phone can do proper supervised transfer, but _only_ once. If the user attempts
to transfer a second time, it won't work.
any s
On Sat, 2005-03-12 at 07:42, Luki wrote:
> Firdosh,
>
> there were couple typos on my last email, but that's essentially what
> I said. There are two ways of doing it -- but neither will work given
> you current setup.
>
> 1) Phone A talks directly to B.
> 2) Both Phone A and B talk to a common p
Hi,
I am again running with presence problem in asterisk.
I have two windows messengers registered successfully
with asterisk (Example msn1 & msn2).
When msn1 adds msn2 in contacts it shows online. Its
fine. But when msn2 un-registers still msn1 displays
msn2 as online (but it MUST be offline)
Is there any application that actually work like Background, but instead of
playing a specified file, it plays the streaming music from music on hold?
the reason i am asking this because i come across a dialplan that goes this
way,
if a person gets to an extension that is busy, it will playback
I am just getting into my Asterisk adventure. I got a
TDM400P with one FXO and one FXS cards. TDM400P is needed a PCI 2.2 according
to the spec. How do I make sure that the PC I will get, has PCI 2.2. How big
the difference among the PCI express, PCI-X, PCI 2.3 and PCI 2.2. What should I
do
> Is there anything I can do to track down the problem? e.g., is there a
> command in *CLI to read the database record? push a record, see the
> differences, ...
You can try this:
realtime update sippeers allow g729 name 621
That should be the SQL equivalent to "UPDATE TABLE sippeers SET allow =
Dear
all:
hi,
i face some problem in the mysql cdr module. Here is my situation, hope
you all give some comments.
(cdr)
User
--->
Asterisk
-->
gateway (IVR) ---> (B
party)
On Tue, 15 Mar 2005 16:29:53 -0500, C F <[EMAIL PROTECTED]> wrote:
> Your problem with MOH has to do with the fact that Asterisk needs a
> timing source to play back music on hold. So when there is an audio
> stream coming in asterisk can use that incoming stream as a timing
> source. If however si
> I try to see ExtensionStatus (event) when I'm logged on manager. But
> nothing :/
> This is implemented in manager.c. May be I compile my astersik with out
> a parameter ?
You have to use the "hint" priority in your dialplan. Then the
ExtensionStatus will work.
http://www.voip-info.org/w
Doh, I should have kept trying for about 10 minutes longer before I
sent that email, the trick is to ensure you have sox (and possibly
soxmix) installed on the Asterisk box. figured I'd answer my own
question should someone else need the answer or possibly just for the
next guy searching the archiv
I've got automon up and recording calls on demand from information I
found in the list archives, however instead of ending up with one
monolithic file, I've got a -in and -out version of the files in my
monitor directory?
Anyone have suggestions how I could end up with a monolithic file that
does
Hello,
>
> I posted earlier with regards to three way calls and X-Lite, this kind of
> yielded everything I already suspected. However I suspect someone has a
> good working config for connecting a third party to an existing call
> (a-la-skype), or a detailed solution of using MeetMe to achieve t
On Tue, 2005-03-15 at 19:04 -0600, Jon Gabrielson wrote:
> On Tuesday 15 March 2005 06:34 pm, Robert Hajime Lanning wrote:
> >
> >
> > > So, let me see if I am right. You run a support shop? You want your
> > > database to validate your data for you instead of leaving that logic
> > > to
> > > you
The first one is my attempt to get the extensions that are in use not
to ring. The second one actually rings all the extensions regardless of
whether someone is on the phone. The reason I want to change this, is
because if you're on a call and the phone rings, the call quality
degrades.
On Mar
Guys.. I just noticed that my grandstream handytone 286 ata are having
problems with voice cutoffs... We can listen to the person on the zap
channel (x100p cards) without problems but they sometimes listen to us with
cutoffs.. like "He ...lo. ow...r.. you" and it comes and goes.. this
doesnt ha
I have setup flash pannel, ... looks nice, but so far I could not
configure it to get more than 4x7 buttons.
I tried to make the buttons smaller, but than just the entire picture is
smaller.
The description says you can have a hundred buttons,
Can I have multiple flash pannels? E.g. for eac
On Tuesday 15 March 2005 06:34 pm, Robert Hajime Lanning wrote:
>
>
> > So, let me see if I am right. You run a support shop? You want your
> > database to validate your data for you instead of leaving that logic
> > to
> > your application? Usually, a database is considered to be an asset
> > wor
Ronald Wiplinger wrote:
Matthew Boehm wrote:
INSERT INTO sip_buddies VALUES
(1,'621',NULL,NULL,NULL,'\"Demo\",<621>','yes','inhouse',NULL,'rfc2833',NULL
,N
ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1',
''
,'999',NULL,NULL,NULL,'Password','fri
Oopps,
sorry Paul I didn't understand your issue, that's for sure. :(
Hmm Interesting thing though.
I'll try it and I'll let you know.
Although how can you reinvite a PSTN line?
They probably have canreinvite=no or similar (because they are not using *) for
billing purposes.
If there is a rein
I've been beating my head a bit against the 1.0.6 Debian builds of
Asterisk, using an E100P (E1, single span) board.
In machines I've built in the past (back in 1.0.0 days), config I'm
using and that card and 1.0.0 driver combo worked fine.
ztcfg reports no problems:
SPAN 1: CCS/HDB3 Bui
Which one did you try? the first one or the second?
what was the CLI output?
On Tue, 15 Mar 2005 18:40:29 -0600, Derrick Shoemake
<[EMAIL PROTECTED]> wrote:
> We have a small number of phones, when a call comes in we want all the
> phones that aren't in use to ring.
> Is there a simple way to tes
Scheda wrote:
Whenever I try to install [EMAIL PROTECTED], I get this error at about 43%
There was an error installing rpmdb-redhat-3.4-0.20050105. This
can indicate media failure, lack of disk space, and/or hardware
problems. This is a fatal error and your install will be aborted.
Please veri
Yeah, It's a 20 gigabyte maxtor drive, so plenty of space.
On Tue, 15 Mar 2005 17:10:49 -0800, snacktime <[EMAIL PROTECTED]> wrote:
> On Tue, 15 Mar 2005 16:17:09 -0800 (PST), beonice <[EMAIL PROTECTED]> wrote:
> >
> > --- Scheda <[EMAIL PROTECTED]> wrote:
> > > Whenever I try to install [EMAIL P
Shane Dalgleish wrote:
I've put my fire proof suit on ready to ask a question ;o)
Redhat 9
Kernel 2.4.20-8
AVM Fritz PCI
Problem 1...
i have followed instructions from:-
http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+
On Tue, 15 Mar 2005 16:17:09 -0800 (PST), beonice <[EMAIL PROTECTED]> wrote:
>
> --- Scheda <[EMAIL PROTECTED]> wrote:
> > Whenever I try to install [EMAIL PROTECTED], I get this
> > error at about 43%
> >
> > There was an error installing
> > rpmdb-redhat-3.4-0.20050105. This
> > can indicat
There is some serious work happening with the Asterisk Attended transfer - the
latest version are getting better, but not without hiccups.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jer
Sent: Tuesday, 15 March 2005 7:00 PM
To: asterisk-users@lists.d
We have a small number of phones, when a call comes in we want all the
phones that aren't in use to ring.
Is there a simple way to test and see what phones are in use then ring
the other phones? I tried some
code like this:
[zap]
exten => s,1,Answer
exten => s,2,ChanIsAvail(${DERRICK})
exten =>
On Tue, 15 Mar 2005 18:52:00 -0500, Giudice, Salvatore
<[EMAIL PROTECTED]> wrote:
> So, let me see if I am right. You run a support shop? You want your
> database to validate your data for you instead of leaving that logic to
> your application? Usually, a database is considered to be an asset wort
> So, let me see if I am right. You run a support shop? You want your
> database to validate your data for you instead of leaving that logic
> to
> your application? Usually, a database is considered to be an asset
> worth
> protecting from unvalidated user input. Also, do you routinely try to
>
--- Scheda <[EMAIL PROTECTED]> wrote:
> Whenever I try to install [EMAIL PROTECTED], I get this
> error at about 43%
>
> There was an error installing
> rpmdb-redhat-3.4-0.20050105. This
> can indicate media failure, lack of disk space,
> and/or hardware
> problems. This is a fatal error and
Folks,
I'm trying to slow down the speed at which Asterisk
decides I've finished typing in an extension for
forwarding voicemail. I've tried using
exten => s,2,DigitTimeout(5);
exten => s,3,ResponseTimeout(5) ;
in my extensions.conf, but it still seems only about 2
seconds (or less!) before
I've recently setup Asterisk. Calls are routed to Asterisk through a
Cisco 1760 router. Some calls originate at Cisco 7960 phones connected
to the router, some originate at other phones that are switched by a
legacy PBX.
My problem is that calls that begin at the 7960s do not seem to transmit
DT
Let's get back to the highly valid point that Robert made. He is
absolutely right. Good parents teach their children to do the right
thing regardless of what the majority of children are doing these days.
Consider 2 approaches to dealing with a compromised PC. I think the
majority of the so-cal
There is always the google cache if you need something specific. Here's
a link to the cmd playback page for instance in text:
http://64.233.161.104/search?q=cache:NwtOSiUWWzUJ:www.voip-info.org/wiki
-Asterisk%2Bcmd%2BPlayback+voip-info+cmd+playback&hl=en&lr=&strip=1
-Original Message-
F
So, let me see if I am right. You run a support shop? You want your
database to validate your data for you instead of leaving that logic to
your application? Usually, a database is considered to be an asset worth
protecting from unvalidated user input. Also, do you routinely try to
insert text stri
Whenever I try to install [EMAIL PROTECTED], I get this error at about 43%
There was an error installing rpmdb-redhat-3.4-0.20050105. This
can indicate media failure, lack of disk space, and/or hardware
problems. This is a fatal error and your install will be aborted.
Please verify your media
Dean,
Appreciate the feedback on the Oz situation.
To be honest maintaining accurate reach list descriptions is pretty hard,
although we're trying to do the best job.
Essentially most of the input to the termination list comes from our Tier 1
interconnects where we get a full dump of their suppo
On Tue, 2005-03-15 at 16:05 -0700, Zanzamar Majere wrote:
> Is anyone else having issues pulling up voip-info.org?
There's been a 'wiki down' thread running all day on this list. So it's
been noticed, yes.
Regards, Bruno.
___
Asterisk-Users mailing
Nope that wont work.
The whole reason you need to change and unchanged mute for conference
calls is when on a cell phone or listening in on 'broadcast' conference
where the majority of the time I will be just listening in eg - a
brokers report or similar but I may want to ask a question at the end
Hi,
>Your problem with MOH has to do with the fact that Asterisk needs a
>timing source to play back music on hold. So when there is an audio
>stream coming in asterisk can use that incoming stream as a timing
>source. If however silence suppression is used, then asterisk has no
>timing source the
Is anyone else having issues pulling up voip-info.org?
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Maybe, it has absolutely nothing to do
with performance or stability. Maybe, it has something with ‘ease of implementation’,
‘ease of use’, ‘availability of commercial support’,
and which database vendors ultimately decide to support in their products. Obviously,
Microsoft has a lot of vend
Giudice, Salvatore wrote:
MySQL: Speed, Power and Precision
Thanks, I will file this in my MySQL Appointment Book under Feb 31.
Oh, you mean that is not a valid date? MySQL had no problem with it...
Seriously though, precision and accuracy are not strongpoints of MySQL.
MySQL really has be
Hi!
> > I couldn't find, for example, a variable containing the current conference
> > name.
> >
> > If I had those I agree it would be simple in the dialplan; just listen for a
> > key eg 2, then when pressed kick user from conference, and immediately
> > rejoin using a mute option, rejoining th
Comment:
"Best" sometimes gets fuzzy.
MySQL's "mind-share" is frightening.
Because of the mind-share/marketing I see MySQL being deployed where
perhaps PostgreSQL should be and Oracle is considered too expensive.
(avoiding MS SQL server. :) )
Also probably due to the 'mind-share' documenation
TC wrote:
Is there any development ongoing with ICD ? I wouldn't want to get
there is on going dev
I'm glad to hear it. Is there a mailing list anywhere that I can
subscribe to ?
involved in something that is not going to make it into CVS HEAD.
and it will never go to cvs head of asterisk :)
W
Sean Kennedy wrote:
As the title suggests, I was wondering if there was another source of
documentation for Asterisk.
Related: If one wanted to contribute to documentation, who would one
contact?
The Asterisk Documentation Project comes to mind:
http://www.asteriskdocs.org
--
Cheers,
Matt Ridde
And it will give you more flexibility for future development.
Having sold pabx/call centre technology in the past I'm still blown away
with how good Asterisk is on an even price basis, the fact that Asterisk
is 1/3rd at the most the price leaves everything else for dead.
Cheers,
Dean
-Orig
Hello,
I am using voipuser.org service, and am trying to make a SIP call.
Everything seems to work fine, except I can't hear anything on my end.
When I make a SIP call, the other party can hear me, but I can't hear
anything. I am using asterisk + Digium TDM board with an FXO port
where
Alternatively you could get someone to custom develop a solution for
you. Believe me the prices are not as bad as you may initially think!
:)
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.
As the title suggests, I was wondering if there was another source of
documentation for Asterisk.
Related: If one wanted to contribute to documentation, who would one
contact?
Thanks!
Sean
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Adam,
I can only comment on Australia but you list one.tel as one of the
companies you can deliver sms's to.
One.tel ceased operations at least 18 months ago if not more.
I'm glad you developed this product and think any external asp delivered
service for asterisk is an exciting development but I
On Tue, 15 Mar 2005 14:50:38 -0700
Daniel Webb <[EMAIL PROTECTED]> wrote:
On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk
wrote:
Dude, where have you been? This has been discussed here
at length.
Everyone agrees that it's on LiveVOIP's end, but they're
shrugging their
shoulders and pointin
cdr_addon_mysql doesn't compile at all, no matter what the OS. I
contacted the authors a while back and they said they would get into
contact with Mark or something... Who knows what happened, but as far
as I know it's still broken.
I tried to compile addons on an x86_64 Opteron under FC3. T
Try googling:
QUERY: "Asterisk-Users" Search String
Works quite well.
--
www.justfuckinggoogleit.com
On Tue, 15 Mar 2005 14:50:38 -0700, Daniel Webb <[EMAIL PROTECTED]> wrote:
> On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote:
>
> > Dude, where have you been? This has been discussed
Thanks Ian for mentioning our offering.
We actually finished the beta yesterday and put the service live today.
Commercial pricing is now up on the website.
As I'm treading dangerously into the waters of self-promotion on a
non-commercial list, I shall say no more, but simply point you towards th
On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote:
> Dude, where have you been? This has been discussed here at length.
> Everyone agrees that it's on LiveVOIP's end, but they're shrugging their
> shoulders and pointing toward *. Search the list.
Could you point out the best way to "sear
Your problem with MOH has to do with the fact that Asterisk needs a
timing source to play back music on hold. So when there is an audio
stream coming in asterisk can use that incoming stream as a timing
source. If however silence suppression is used, then asterisk has no
timing source therefore MOH
> The only two things I have found that doesn't work is a) the mailbox
> entry for a SIP user doesn't actually light up the MWI (Message
> Waiting Indicator); and b) voicemail passwords cannot begin with a
> '0' (zero) because its a numeric field.
You are behind the times. The MWI now works. Y
OK. I just grabbed libpri and zaptel via:
cvs co libpri zaptel
I compiled and installed and was unable to make outbound calls. Inbound was
fine.
(See my previous post about call length 0). The changelog in both dirs
showed the version as 0.1.6.
This "should" be the newest version of libpri and za
Matt Schulte wrote:
> anything, driving me nuts. I'm running asterisk 1.0.6, as head won't
"Take your 'blah-blah' to the 'blah-blahtologist'." - Dr. Cox, Scrubs
RealTime requires CVS-HEAD! That is why its not working with 1.0.6!!!
Perhaps I should make the font on the wiki larger..hmm..
-M
Eric,
many thanks for your prompt reply.
>> Using X-Lite to dial extension 400, I hear it ring and then get
>>answered
>> and I hear about 0.1 of a second of the on hold music and then silence.
>> If I use the 'line 1' button to put the call on hold and then take it
>> off again I hear a
Title: AntiSpam Alert: Request For Authentication
I wondered where that came from. I just deleted it and continued.
On Tue, 2005-03-15 at 12:04, dean collins wrote:
Is there anyway we can get this shit off the asterisk list apart from posting their email address [EMAIL PROTECTED] here fo
> -Original Message-
> From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> > Top Deployed Databases poll shows following databases in use:
> >
> > SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%.
>
> I see they created this with Mysql,
> 78 + 55 + 44 + 8 = 185%
> I'
On Tue, 2005-03-15 at 14:21 -0500, Giudice, Salvatore wrote:
> Sticks and stone still break my bones, but PostgreSQL is still a dog.
>
>
>
>
> Market share:
> According to CD Times magazine dated July 1, 2004
>
> Top Deployed Databases poll shows following databases in use:
>
> SQL Server
Does anyone know the answer to this question? It's causing me endless
frustration.
tony
Zero Effort Networking
Pty Ltd ABN 38 082 434 446
PO Box 6045
Blacktown NSW 2148
www.zeroeffortnetworking.com.au
[EMAIL PROTECTED]
Tel: (02) 9676 3541
Fax: (02) 8569 2012
Jed Stafford wrote:
> Just to be safe, I removed everything and got a fresh
> copy from CVS. Again I got Asterisk to compile fine,
> but the addons do not. Just to double check you got
> this to compile on OS X? Here is what I get.. I did
> change from static to dynamic which seems to get
> things
Kanishka,
For this question, all your previous questions and
possibly all your future questions, you have to search google
first or you have to find a consultant, who will help you get
going.
No one can give you a clear and specific answer for a
general question like ' I do not get the
Neil A. Hillard wrote:
Using X-Lite to dial extension 400, I hear it ring and then get answered
and I hear about 0.1 of a second of the on hold music and then silence.
If I use the 'line 1' button to put the call on hold and then take it
off again I hear another 0.1 of a second of the music. This
Hi,
my asterisk is connected to the internal S0 of my ISDN PBX. Dialing
20 with a phone connects to the HFC-S card in my asterisk just fine,
I get a dialtone and can get out via SIP. So far so good. I would like
to be able to dial 20 without a wait, though. With i4l DTMF,
this worked fine (if DTMF
On Mar 11, 2005, at 7:20 PM, Scott Laird wrote:
On Mar 11, 2005, at 5:31 PM, TC wrote:
I was just reading the release notes for the latest SPA-841 firmware,
and noticed that Sipura added support for "SIP-B" to this release.
This apparently adds support for bridged line appearances, parking
softkeys
Hi,
I've recently installed Asterisk and have got the majority of it
configured (what an excellent piece of software it is, too), but I'm
having a couple of problems.
The first one is with music on hold! I've downloaded and
installed mpg123 as specified:
# whereis mpg123
mpg123: /u
I could start a pretty big flame war if I tried to compare Windows 95 with
MacOS X by deployment stats instead of stability. [David
Brodbeck] I've seen Mac OS X locked up solid just by putting in
a damaged CD-R disc. It's a nice OS, mind you, but it's not as
stable as some peop
From my personal experience, pgsql outperforms mysql when using tables
with over 30.000.000 records.
For small tables mysql is faster, but also locks up more when 1 thread
takes a long time.
We used mysql for years, then had to move on to pgsql and never turned
back. (we still have some 300 queries
has any one installed this, i just tried this on a
test server, i get voice but it's corrupted, i do not get the natural
voice
any idea why
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Kanuri, Seshu (Company IT) wrote:
Thanks for the pointers. Here is my Features.conf where I have tried my
best to use Asterisk to give away control. I have enabled ## as the
combination key for Asterisk (in quick succession) to retain control,
but otherwise ignore the key presses.
I don't run CVS-H
On March 15, 2005 02:21 pm, Giudice, Salvatore wrote:
> Sticks and stone still break my bones, but PostgreSQL is still a dog.
Until you actually show some benchmarks where the tests are clearly documented
and Postgres is properly tuned, you're spreading FUD. Your testing should
also demonstrate
On Mar 15, 2005, at 11:21 AM, Giudice, Salvatore wrote:
Sticks and stone still break my bones, but PostgreSQL is still a dog.
Market share:
According to CD Times magazine dated July 1, 2004
Top Deployed Databases poll shows following databases in use:
SQL Server with 78%, Oracle - 55%, M
Marios,
I don't think you quite understand my issue. The ata in my apartment is
behind a nat, and it always has had canreinvite=no. But my question
deals not with my sipura device, but calls entirely contained within the
* server (which again, is a live IP machine). A call comes in from
broadvoice
On Tue, 2005-03-15 at 20:09 +0100, Andreas Meyer wrote:
> Sorry for not being clear enough but my headphone is attached to the
> soundcard at my local PC. Now when I start Asterisk on that machine it
> is using port 5060 and sjphone can not connect because it also uses port
> 5060.
>
> netstat -p
Giudice, Salvatore wrote:
Sticks and stone still break my bones, but PostgreSQL is still a dog.
Enough, take it off list, PLEASE!
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If you want to authenticate by IP, you need to add: insecure=very
Julian J. M.
On Tue, 15 Mar 2005 17:19:17 -, Kanishka Somaratne
<[EMAIL PROTECTED]> wrote:
> I want to enable SIP calls from an ip address, direct calling without
> registering, the ip which sends the calls will not change. i
Have you considered using the mysql method instead of the odbc method. I'm
using it and it works just fine. Here's a sample of my extconfig.conf:
extensions => mysql,ast-conf,extension
sipfriends => mysql,ast-conf,sip_buddi
voicemail => mysql,ast-conf,voicemail
You also need to add the
Sticks and stone still break my bones, but PostgreSQL is still a dog.
Market
share:
According to CD Times magazine dated July 1, 2004
Top Deployed Databases poll shows following databases in use:
SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL -
8%.
--
Eric Wrote:
---
The trick is not to use options you don't understand. "show application
dial" will show you what the t and T options are for.
Most people use the transfer feature of their phone, rather than using
the T/t hack on the Dial line.
Sounds like you are using CVS-HEAD and so wi
On Mar 15, 2005, at 10:40 AM, Steven Critchfield wrote:
Won't help as the software he is using specifically checks the from:
header. Just another example of broken methods to fix this problem.
Also
it is unlikely he can fix the software as it is a commercial solution.
It's pretty lame that some AS
"Giudice, Salvatore" <[EMAIL PROTECTED]> wrote:
> Do you have a soundcard? If so, yes.
> I am new to the list and got a simple question. Is it possible to connect
> with a headphone to Asterisk on the same machine?
> I tried a portrange 5060:5061 in sip.conf but that doesn't work. I can
> connect
> Is there any development ongoing with ICD ? I wouldn't want to get
there is on going dev
> involved in something that is not going to make it into CVS HEAD.
and it will never go to cvs head of asterisk :)
>
it will go to stable of http://aefirion.org/ & replace app_q chan_agent
> I really would
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