Brian West wrote:
You can't anymore MySQL was ripped from Asterisk because the client libs
are GPL.
bkw
On Tue, 30 Sep 2003, Manoj K Gupta wrote:
I still don't get this.. Asterisk is GPL (with an option of a commercial
licence), MySQL is GPL(with an option of a commercial licence on the
ser
Jamie Carl wrote:
All registered with Sourceforge. The new project will be called
'AstWeb' and will be available under the GNU General Public Licence.
The current release of AstCDR will be ported over in the next day or
so along with the addition of a few features and labelled 'AstWeb v0.4'.
On Tuesday 30 September 2003 01:01, Brian Capouch wrote:
> I wonder if I'm the only one who finds Allison reading the timestamp
> on my voicemails in GMT, after months of having it done with the
> local time I have set in voicemail.conf.
>
> The timestamps on the message files, and the emails that
Mark Evans wrote:
I think we're getting away from the original purpose of this program.
Are people really that desparate for a full, web-based admin/user
interface?
I sure am, I want to give as much control as I can for basic tasks
to my customer who may not even know what Linux is :)
Regard
I wonder if I'm the only one who finds Allison reading the timestamp on
my voicemails in GMT, after months of having it done with the local time
I have set in voicemail.conf.
The timestamps on the message files, and the emails that are sent, are
correct. Only the spoken dates appear to be affe
On Mon, 2003-09-29 at 13:23, Jeff Dodge wrote:
> So -- If you don't distribute the compiled app to me -- I have no right to
> ask you for the source. Even if I pay
> you for your custom application and you must provide me with the source
> (Upon request!) I have no redistribution rights
> to that
You can't anymore MySQL was ripped from Asterisk because the client libs
are GPL.
bkw
On Tue, 30 Sep 2003, Manoj K Gupta wrote:
> Hi list,
>
> I am trying a scenerio where the * will take the email and mailbox number from the
> Mysql database for sendming mail to a voicemail user. I have seen v
> The M phones from Nortel are digital phones as used with Norstar or
> Meridian 1 systems.
Actually some if not all M8XX and M9XX phones, the 9516 for example (which are
now sold by Aastra) are just analog phones with lots of buttons and stuff...
they do work nicely with asterisk.
Brad
__
http://www.loligo.com/asterisk/current/
I'm sure he has a few in his sip.conf examples
On Mon, 29 Sep 2003, Leif Madsen wrote:
> Hi all,
>
> I would like people to email me at 'leif at hacklocalhost dot com' some
> example configuration files for VoIP providers which * can register
> with. I am
Hi list,
I am trying a scenerio where the * will take the
email and mailbox number from the Mysql database for sendming mail to a
voicemail user. I have seen vmdb.sql file but is not able to determine its
use.
I have created a table using that file but don't
know how to use the fields fr
All registered with Sourceforge. The new project will be
called 'AstWeb' and will be available under the GNU
General Public Licence. The current release of AstCDR
will be ported over in the next day or so along with the
addition of a few features and labelled 'AstWeb v0.4'.
The project is cu
Hi:
I have the next problem at VoiceMail application, when i call is
recive, it does not detect when the user hangup the call..
The first solution i think was to place the busydetect=yes at
zapata.conf but it cause that some calls get lost ( when some one is
talking, the * hangup the call).
> I think we're getting away from the original purpose of this program.
> Are people really that desparate for a full, web-based admin/user
> interface?
I sure am, I want to give as much control as I can for basic tasks
to my customer who may not even know what Linux is :)
Regards
Mark
__
I think that you missed my point. I am not proposing to establish a forum
and abolish the maillist.
The forum would get traffic as all posts sent to the maillist would
automatically post to the forum. Those that want to answer using the forum
can do so and it would forward to the list.
If
Jamie Carl wrote:
I think we're getting away from the original purpose of this program.
Are people really that desparate for a full, web-based admin/user
interface?
If so, tell me, and we'll make this an official project rather than just
some code I slapped together in an hour. :)
I think it'
I have a client near Budapest that is looking to get two X100P cards.
They are looking for someone close by that can ship to them in 1 to 2
days. (as in, they would receive the card within 2 days of shipping.)
I will be remotely setting up a small Asterisk server there.
--
END OF LINE
_
Anyone willing to try and call me?. Or is there "test" IP's to tryout a
newer phone?
My address is 142.59.179.203 and the phone is configured as point to point
running h323.
Craig
___
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[EMAIL PROTECTED]
http://lists.digium
Hi list,
If I am using a T1 span to link with another PBX. for a particular call,
after the called party answers the call, how can I identify which channel of
the span is used for the call ??
The deplyoment is as follows:
caller---Asterisk T1--- PBX---called party.
Assume I am using th
Hi all,
I would like people to email me at 'leif at hacklocalhost dot com' some
example configuration files for VoIP providers which * can register
with. I am going to expand upon the FWD php "wizard" I created for
these other providers, but I need some examples as I don't actually use
anythi
Thanks. That worked
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thorsten
Lockert
Sent: Tuesday, 30 September 2003 11:43 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: SIP i.e. Is something broken?
To roll back only the affected stuff for
Leif Madsen wrote:
I'm hoping someone can help me out with this. I am basically just
trying to separate my outbound and inbound calls into separate contexts
instead of having everything in a single context. Any help would be
appreciated. Perhaps I've missed something really obvious
Yah,
Uriel Carrasquilla wrote:
Leif:
I take it that the * on the GW machine remains in the middle of the call
between SIP and the Nat'ed *, correct?
I also suspect that you will be using IAX over Ethernet (to avoid
compression/decompression delays), correct?
Great work!
You'd be correct. IAX is over Et
If I may, I'd like to make a few points that I hope can help find an answer
to what we seem to be looking for: THe equitable distribution of financial
gains due to efforts brought to the table by Digium, efforts to contributors
to Asterisk and the efforts of independent operators.
Analogy:
In the
Why couldn't we just use PEAR?
it is there, it works and it does provide the abstraction layer for any of
the RDMBS systems discussed so far.
URiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Evans
Sent: Monday, September 29, 2003 11:10 AM
To: [EMAIL
To roll back only the affected stuff for SIP negotiation, I would recommend:
make update
cvs update -j 1.181 -j 1.179 channels/chan_sip.c
Note that the second line should only be executed *once*. Once this is
fixed
in CVS, you should *remove* channels/chan_sip.c to make sure the
On Mon, 29 Sep 2003 22:08:28 -0400
"Uriel Carrasquilla" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
How about including VoiceMail viewer/retriever.
URiel
I think we're getting away from the original purpose of
this program. Are people r
On Mon, 29 Sep 2003, WipeOut wrote:
> Dave Weis wrote:
> >On Mon, 29 Sep 2003, Brian Capouch wrote:
> >>Christopher J. Wolff wrote:
> >>>Is it safe to assume that a fresh CVS build will not have the SIP
> >>>translation problem described?
> >>>Just FYI: I had similar problems for a while, and then
How about including VoiceMail viewer/retriever.
URiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Evans
Sent: Monday, September 29, 2003 5:37 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CDR Web Search Frontend
> Or do something really sm
Leif:
I take it that the * on the GW machine remains in the middle of the call
between SIP and the Nat'ed *, correct?
I also suspect that you will be using IAX over Ethernet (to avoid
compression/decompression delays), correct?
Great work!
Regards,
Uriel
-Original Message-
From: [EMAIL PRO
Force a module not to load (mv file), and you get this.
WARNING[8192]: File loader.c, Line 347 (load_modules): Loading module
cdr_mysql.so failed!
[EMAIL PROTECTED] sbin]# Ouch ... error while writing audio data: : Broken
pipe
MA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I am a fairly new user of Asterisk and I am generally impressed with its
features. I have some questions about the SIP channel support:
1. I have noticed that even when there are no active calls, there is a list of
active SIP channels. This appears to be a bug. Has anyone seen this?
Yes. Bug
I filed a bug report yesterday about it.
http://bugs.digium.com/bug_view_page.php?bug_id=330
Budgetones are effected, not sure about others. It seems to be codec
related. If you use allow=all, then it tries to negotiate G723 with Ulaw
and this effects other audio items.
MA
-Original Me
cdr_generic? :P
On Tue, 30 Sep 2003, Jamie Carl wrote:
>
>
> Guys! I'm putting the source up on SourceForge on my
> existing account. Questions is this tho:
>
> What should we call it?
>
> Is AstCDR good/fancy enough?
>
> Suggestions please! I would like to get this on SF by the
> end of the d
>> Is AstCDR good/fancy enough?
How about AstManager as I think this should do more than just the CDR
part and get involved in configuration/management
Reporting and also user management.
Just my 2 cents though :D
Regards
Mark
___
Asterisk-Users m
Guys! I'm putting the source up on SourceForge on my
existing account. Questions is this tho:
What should we call it?
Is AstCDR good/fancy enough?
Suggestions please! I would like to get this on SF by the
end of the day. (it's 9:33am here).
Someone I know once said:
"I'm not a coder,
WipeOut wrote:
Lists wrote:
This is my issue as well, Does anyone know how to fix it?
Roll back to the CVS from last Thurdsay, This worked for me..
If you like you could try Friday and see if it works which will help
narrow down when the problem started.. :)
I'm going to bet that it's codec ne
I went back to thrusdays CVS when I call a zaptel interface from the sip
the sound on the ZAP side is very choppy (what is being said by the BT-101
user) However what is said on the zap side (and heard by the BT user) is
nice and clear.
The BT phone is on a LAN with the asterisk box so there is no
Brian West wrote:
I have found the CDR in general to be a problem, We use a system that
allows a user to simply click a phone number in a web page and then PHP
drops a call file into the /outgoing directory.. These calls are not
logged at all.. not in the text file or the MySQL..
Can I make c
> I have found the CDR in general to be a problem, We use a system that
> allows a user to simply click a phone number in a web page and then PHP
> drops a call file into the /outgoing directory.. These calls are not
> logged at all.. not in the text file or the MySQL..
Can I make calls thru your
> Why do they do that? Quite possibly because they, like myself, hate
> having to scroll through pages and pages of quotes to get to the reply,
> which isn't always clear where it might start.
Well I prefer to know to what you refer to when you send a reply to a
messgae. I don't want to read the
Andres wrote:
On Monday 29 September 2003 17:02, Brian West wrote:
ie a perl script. :)
On Mon, 29 Sep 2003, Mark Evans wrote:
As far as I am aware the CDR logging can only log
to MySQL or CSV text file..
Unfortunately the CSV Logger is not that reliable. I have found times wh
Lists wrote:
This is my issue as well, Does anyone know how to fix it?
Thanks,
Michael
Roll back to the CVS from last Thurdsay, This worked for me..
If you like you could try Friday and see if it works which will help
narrow down when the problem started.. :)
Later..
___
On Monday 29 September 2003 17:02, Brian West wrote:
> ie a perl script. :)
>
> On Mon, 29 Sep 2003, Mark Evans wrote:
> > >> As far as I am aware the CDR logging can only log
> > >> to MySQL or CSV text file..
Unfortunately the CSV Logger is not that reliable. I have found times when it
has stop
Dave Weis wrote:
On Mon, 29 Sep 2003, Brian Capouch wrote:
Christopher J. Wolff wrote:
Is it safe to assume that a fresh CVS build will not have the SIP
translation problem described?
Just FYI: I had similar problems for a while, and then I completely
scrapped my CVS directory and did a C
This is my issue as well, Does anyone know how to fix it?
Thanks,
Michael
On Mon, 29 Sep
2003, Dave Weis wrote:
>
> On Mon, 29 Sep 2003, Brian Capouch wrote:
> > Christopher J. Wolff wrote:
> > > Is it safe to assume that a fresh CVS build will not have the SIP
> > > translation problem descri
Brian West wrote:
ie a perl script. :)
Or a PHP script..
PHP from the console works great and the DB connection stuff is usually
there.. :)
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
> Actually, top posting, and yes, people do that.
>
> Why do they do that? Quite possibly because they, like myself, hate
> having to scroll through pages and pages of quotes to get to the reply,
> which isn't always clear where it might start.
I'll second that motion... proposal passed!!!
Tho
WipeOut wrote:
Your ATA-186 and Bugetone are both behind NAT or both non-NAT??
In otherwords are they both in the same setup in relation to the server??
The setup is identical; both are behind a NAT translating router.
B.
--
This message has been scanned for viruses
and is believed to be cle
On Mon, 29 Sep 2003, Brian Capouch wrote:
> Christopher J. Wolff wrote:
> > Is it safe to assume that a fresh CVS build will not have the SIP
> > translation problem described?
> >
> > Just FYI: I had similar problems for a while, and then I completely
> > scrapped my CVS directory and did a CVS
ie a perl script. :)
On Mon, 29 Sep 2003, Mark Evans wrote:
> >> As far as I am aware the CDR logging can only log
> >> to MySQL or CSV text file..
>
> It doesn't matter, we could write an automatic import routine which can
> import that CVS into anything :)
>
> Regards
>
> Mark
>
>
> ___
Actually, top posting, and yes, people do that.
Why do they do that? Quite possibly because they, like myself, hate
having to scroll through pages and pages of quotes to get to the reply,
which isn't always clear where it might start.
With top posting, you know the reply starts at the top, and
Mark Evans wrote:
As far as I am aware the CDR logging can only log
to MySQL or CSV text file..
It doesn't matter, we could write an automatic import routine which can
import that CVS into anything :)
Regards
Mark
___
Asterisk-Users mailing list
The M phones from Nortel are digital phones as used with Norstar or
Meridian 1 systems. As such, they won't work with the TDM400P card.
The Aastra 390 phone is an ADSI analog phone that works well :-)
Cheers
Paul
___
Asterisk-Users mailing list
[EM
I doubt if you'll be able to. They made a strategic decision to GPL
rather than LGPL their client access libraries, as they wanted to up
their proprietary license revenue. Essentially, they're trying to
enhance the benefits of paying for a commercial license fee, by making
it difficult to use MyS
>> As far as I am aware the CDR logging can only log
>> to MySQL or CSV text file..
It doesn't matter, we could write an automatic import routine which can
import that CVS into anything :)
Regards
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED
Brian Capouch wrote:
Christopher J. Wolff wrote:
Is it safe to assume that a fresh CVS build will not have the SIP
translation problem described?
Regards,
Christopher
--__--__--
Just FYI: I had similar problems for a while, and then I completely
scrapped my CVS directory and did a CVS CHECKOUT
Nope...
Nortel's M series is for the Meridian series PBXs/switches... It's a
proprietary digital interface.
I doubt it will ever happen.
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason A.
Pattie
Sent: Monday, September 29
Hi all,
I am having a problem detecting a FAX machine on the other end when I
dial out a number, even though I hear it playing the 'violin' when it
answers. I would like to be able to detect the event of having a FAX
answering the call I generate through Asterisk. I am testing on a X100P.
I looked
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> I've searched the mailing list quite extensively, but didn't come up
> with anything promising (some things wer helpful, though). Does anyone
> know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310)
> phones can be made to work w
> Personally, I *love* MySQL, and I'm a bit surprised by their sudden change
> from public domain (and maybe LGPL) to GPL for their client libraries...
Who can we bug at mysql to see if we can get that changed?
bkw
___
Asterisk-Users mailing list
[EMAIL
On Monday 29 September 2003 15:55, Mark Spencer wrote:
> > As far as I am aware the CDR logging can only log to MySQL or CSV text
> > file..
> >
> > Also does anyone know how this issue with Asterisk and MySQL is going to
> > affect things like the MySQL CDR logging?
>
> MySQL CDR logging is incomp
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've searched the mailing list quite extensively, but didn't come up
with anything promising (some things wer helpful, though). Does anyone
know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310)
phones can be made to work with the TDM400
Christopher J. Wolff wrote:
Is it safe to assume that a fresh CVS build will not have the SIP
translation problem described?
Regards,
Christopher
--__--__--
If you do try the current CVS and UA's behind NAT work please let me know..
later..
___
Aste
> I am surprised to the opposition to forums. Are we living in the stone
> age? I have used many forums and find them 100 times easier than mailing
> lists. Imagine having to get emails from 10's of forums filling up my
> mailboxes every day.
Web forums suck ass, that is why. They're much slower
Christopher J. Wolff wrote:
Is it safe to assume that a fresh CVS build will not have the SIP
translation problem described?
Regards,
Christopher
--__--__--
Just FYI: I had similar problems for a while, and then I completely
scrapped my CVS directory and did a CVS CHECKOUT (instead of an update).
> As far as I am aware the CDR logging can only log to MySQL or CSV text
> file..
>
> Also does anyone know how this issue with Asterisk and MySQL is going to
> affect things like the MySQL CDR logging?
MySQL CDR logging is incompatible with OpenH323 basically. You now have
an asterisk-addons whe
Hey all,
Does anyone have a reference for the differences between the various US PRI
protocols (Nat'l 1, Nat'l 2, 5E, DMS, etc)? I'm specifically interested in
features available via signalling. I've read that National ISDN, for the most
part, has a smaller subset of features than any other p
> So, if 3rd parties are doing or going to do that, then why not allow
> them to do it in a way that doesn't require bypassing proper design?
> A third party could then for example start selling G.723 codecs, if they
> are prepared to pay the fee that allows them to do so.
We want to promote the a
Is it safe to assume that a fresh CVS build will not have the SIP
translation problem described?
Regards,
Christopher
--__--__--
Message: 11
Date: Mon, 29 Sep 2003 12:45:40 -0700
To: [EMAIL PROTECTED]
From: "Ernest W. Lessenger" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Is somthing broken
On Mon, 2003-09-29 at 19:16, Keith O'Brien wrote:
> I'll offer one better. Why don't we mirror all of the maillist posts to a
> forum. That way both parties are happy. Those that want a forum can use a
> forum interface and still post to the maillist and those that like the
> maillist can stay
Mark Evans wrote:
Hi All
I am happy to go with SF. Jamie do you want to apply for an account
What I was thinking for the DB layer is the following
We have a main class which contains a basic DB implementation. We then
create subclasses which extend the main class for each DB we want
support
Hi,
On Mon, 2003-09-29 at 16:40, Mark Spencer wrote:
> > 1) if your application is not released to a 3rd party, you do not have
> > to make the source available
>
> This is TRUE.
>
> > 2) if you build your application as a module that loads into a stock
> > asterisk server, you do not have to di
At 12:33 PM 9/29/2003, you wrote:
Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram)
Just FYI: I had similar problems for a while, and then I completely
scrapped my CVS directory and did a CVS CHECKOUT (instead of an update).
That solved the problem.
--Ernest
Mark
On Mon,
Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram)
Mark
On Mon, 29 Sep 2003, WipeOut wrote:
> WipeOut wrote:
>
> > Hi,
> >
> > I updated my live server yesterday(after testing on my Dev server
> > first, all works on the Dev server)..
> >
> > Here is the setup..
> >
> > SIP_
> > > 1) if your application is not released to a 3rd party, you
> > do not have
> > > to make the source available
> >
> > This is TRUE.
> >
> > > 2) if you build your application as a module that loads into a stock
> > > asterisk server, you do not have to disclose your source
> >
> > This is FAL
Red Carpet will give you some serious dependency problems later down the
road.
Bisker, Scott (7805) wrote:
I found the best way to upgrade is install Red Carpet from www.ximian.com.
Subscribe to the RH 9.0 channel. And do a complete update. The only
drawback is that this method doesn't update
It's interesting.. cos I'm a big proponent of message forum/bulletin board
type systems, it's great to see the interaction amongst people etc. And
Costas has some very valid arguments for why/how they provide value. I came
in to this thread early on with a "Hmm, yeah, I like forum systems too"
ment
Actually it's BSD licensed. I'll try to bring it up with them to get the
copyright notice provided.
Mark
On Mon, 29 Sep 2003, Brian West wrote:
> I recall someone saying that hdparm is embeded in the codec and
> Registration binaries.. and that is a violation of the GPL. But thats
> voiceage's
Hi All
I am happy to go with SF. Jamie do you want to apply for an account
What I was thinking for the DB layer is the following
We have a main class which contains a basic DB implementation. We then
create subclasses which extend the main class for each DB we want
supported.
This code woul
> I would also like some more info on this whole mysql being taken out of
> the core asterisk install. I understand its because of the dual lic. that
> digium has.. gpl and comercial... why can't mysql be non-existant in the
> comercial version. Then mysql would be compatible with asterisk?!? Or
> > > 2) if you build your application as a module that loads into a stock
> > > asterisk server, you do not have to disclose your source
> >
> > This is FALSE. Even modules for Asterisk MUST be released under GPL,
> > unless you obtain a license to release them outside of GPL from Digium.
>
> You
Leif Madsen wrote:
OK.. lets just simplify this a bit.
<--TDM400P--> <*> <--IAX--> <*> <--SIP-->
I want to make a call from the very left asterisk box, through the right
asterisk box to the remote end. So I have a phone plugged into the left
* box, which is connected via an IAX connection t
costas <[EMAIL PROTECTED]> said:
>7) And then the guy who said the horse can be faster than an automobile.
>Although a romantic notion of the old west Pony Express or something,
>wanna race east coast to west coast against my beat up Chevy?
>
I do. Wanna race a clumsy webbrowser against trn for re
WipeOut wrote:
Hi,
I updated my live server yesterday(after testing on my Dev server
first, all works on the Dev server)..
Here is the setup..
SIP_UA---[NAT]---Asterisk1---PSTN(chan_capi)
The SIP_UA is able to recieve calls from the server with no problems..
Initiated from the PSTN or my Dev
--- costas <[EMAIL PROTECTED]> wrote:
> I would consider GPL my code.
>
> I am just afraid that if SCO bought Digium then they would claim its
> their code and sue everyone. Just because I'm paranoid it doesnt mean
> they are not out to get me. :)
It does not matter who buys Digium, Even if S
> Then you haven't been around Open Source software long, have you.
> There are many "wars" on going some going on many years now.
Yes but I have never paid much attention to all this drama.
bkw
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[EMAIL PROTECTED]
http://l
On Mon, 2003-09-29 at 18:25, Matthew Hardeman wrote:
> I've never attempted this in CIPE, though I have tested a SIP session
> through IPSEC with no trouble.
I've run Netmeeting through openvpn, so SIP should go through OK.
--
Dave Cotton <[EMAIL PROTECTED]>
_
A maybe better sollution would be to replace MySQL with PostgreSQL.
PostgreSQL being covered by a BSD license allows it to be distributed
with Asterisk no mater how Asterisk is licensesd. Check it out at
www.postgresql.org
--- Brian West <[EMAIL PROTECTED]> wrote:
> I would also like some more
Top-quoting. Argh.
On Monday 29 September 2003 12:16 pm, Keith O'Brien wrote:
> I'll offer one better. Why don't we mirror all of the maillist
> posts to a forum. That way both parties are happy. Those that
> want a forum can use a forum interface and still post to the
> maillist and those t
Actually it could be done to a set of newgroups much easier
On Mon, 29 Sep 2003 10:25:25 -0700, Ernest W. Lessenger wrote:
>At 10:16 AM 9/29/2003, you wrote:
>
>>I'll offer one better. Why don't we mirror all of the maillist posts to a
>>forum. That way both parties are happy. Those that
At 10:16 AM 9/29/2003, you wrote:
I'll offer one better.
Why don't we mirror all of the maillist posts to a
forum. That way both parties are happy. Those that want a
forum can use a
forum interface and still post to the maillist and those that like
the
maillist can stay as is.
Would have to
I'll offer one better. Why don't we mirror all of the maillist posts to a
forum. That way both parties are happy. Those that want a forum can use a
forum interface and still post to the maillist and those that like the
maillist can stay as is.
Would have to write a Perl script to pull this
Maybe we should say "RFTM" or in this case "RTFGPL". If is easy to
see how non-GPL modules could be linked to GPL code. Two methods
come to mind:
(1) If the modual were "general purpose" in the sense that it COULD be
used with
something other then Asterisk. In other words the modual does not
I would consider GPL my code.
I am just afraid that if SCO bought Digium then they would claim its their code and
sue everyone. Just because I'm paranoid it doesnt mean they are not out to get me. :)
Costas
-- Original Message --
From: Chris Albertson <
I would imagine that it would boil down to the niceties of having one
source tree for the core product, regardless of license chosen?
Just a guess.
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, Septembe
Mark's right but note that if you write an application for a client,
then that is "distributing" and the provision in the GPL applies.
Only if you keep it to yourself. If you wrote it for use within your
company then you are OK. We must assume that because the person
posting
the question was c
I ment BSD lic. mybad. :P
On Mon, 29 Sep 2003, Brian West wrote:
> I recall someone saying that hdparm is embeded in the codec and
> Registration binaries.. and that is a violation of the GPL. But thats
> voiceage's doing. Anyone care to shed some light on this?
>
> bkw
>
> On Mon, 29 Sep 200
I recall someone saying that hdparm is embeded in the codec and
Registration binaries.. and that is a violation of the GPL. But thats
voiceage's doing. Anyone care to shed some light on this?
bkw
On Mon, 29 Sep 2003, Eric Wieling wrote:
> On Mon, 2003-09-29 at 09:40, Mark Spencer wrote:
>
> >
I would also like some more info on this whole mysql being taken out of
the core asterisk install. I understand its because of the dual lic. that
digium has.. gpl and comercial... why can't mysql be non-existant in the
comercial version. Then mysql would be compatible with asterisk?!? Or am
I wr
I've never attempted this in CIPE, though I have tested a SIP session
through IPSEC with no trouble.
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad
Jordanovic
Sent: Monday, September 29, 2003 2:28 AM
To: [EMAIL PROTECTED]
S
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