Many of you may have seen the recent announcement about Danny Windham
coming on as the new CEO of Digium. This is one of the most exciting
things to happen to Digium and to Asterisk at large. When Danny comes on
board, I will be transitioning to the role of Chief Technical Officer
(retaining
Wanted to let you guys know that Digium has hired Kevin Fleming to assist
me with Asterisk development full time.
Brian West felt it was important for me to reassure everyone that Digium
remains committed to Open Source and of course has no plans to change from
our dual license model in which
The Paris Asterisk meeting will be held Monday, December 20, 2004 at 1
p.m. at Les Vendanges--a wonderful restaurant in the 14th (tel
01.45.39.59.98). However, we have to let them know exactly how many
people will attend, so PLEASE RSVP as soon as possible.
The address is 40, rue Friant, and
I will be in Paris this December and would like to know if anyone would
like to meet to have an informal Asterisk get together. Please e-mail me
directly off-list your availability Dec 19, 20, 21 and 22 and we will
select whichever day is most convenient for the most people. Please try
to
There seems to be some confusion here so I would like to make a few brief
comments and will likely not add much to this thread other than these few
things:
1) Digium *does* license Asterisk (as we distribute it, no additional
features) outside of GPL and we *do* have commercial licensees
I hope this is the last time I have to comment on this...
I'm struggling to think of another free software project where contributed
code bearing an identical GPL or BSD license would require any such
additional disclaimer.
How about GCC, GLIBC, etc? Do you think Digium came up with those
Regardless of whether or not you have licensed G.729 from SIPRO
independently of Digium, the distribution of the codec, linked against
Intel's proprietary IPP library, is clearly and totally in direct
violation of the terms of the GPL. There is no room for argument on this
issue.
We are
To everyone who spends time in #asterisk or #asterisk-bugs or basically
anything with #asterisk in its name, I want to implore you to please treat
new users with respect, and act as good representatives of the Asterisk
community. Recently I have had more reports of new users being severely
We have officially made the first release candidate of Zaptel, Libpri,
Asterisk and Gastman available. While there are still open major bugs,
they are relatively limited, and it was time to go ahead and get the 1.0
ball rolling in earnest.
ftp://ftp.digium.com/pub/asterisk
Enjoy the code.
I am hereby announcing the immediate availablity of ASTCC for *alpha*
testing. ASTCC is an AGI script and CGI script which greatly simplifies
the task of creating a calling card application on Asterisk. Just check
it out of Asterisk CVS as module astcc:
export CVSROOT=:pserver:[EMAIL
Okay, setting aside conspiracy theories, trolling, flaming, etc, let me
summarize some differences between SIP and IAX, and it might help you
make a decision about what is best for you.
1) IAX is more efficient on the wire than RTP for *any* number of calls,
*any* codec. The benefit is anywhere
I will be coming to Paris and then Bordeaux for the Libre Software Meeting
(LSM) 2004. We will have a small get together for Asterisk users in
France or Europe on Friday July 2, 2004 at Le Lateral, 4 Avenue Macmahon
(Metro Charles de Gaulle Etoille, sortie Avenue de Wagram) at 12:00 p.m.
So, in
Not with the voiceage system, but with the new system you will be able to.
Mark
On Sat, 8 May 2004, nicolas wrote:
if i have ordered one lic.
and now i have realized i need two lic for one call (2 cannels one to
provider one to sipphone)
can i install 2. lic with another reg code ?
nico
A Solution now exists for those of you still looking for the first
model of Quad T1/E1 Interface Cards - the T400P and E400P. The T400P
and E400P were manufactured by Digium until a few months ago, and were
designed and released under the GNU Public License as the Tormenta II
by Zapata
I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
Unfortunately - I only seriously scanned the mailing list after buying
the keys
Seems
I've been considering the nature of Asterisk, its security, the bug
tracker, and more... And i've come up with an interesting idea: A
message of the version. The idea is that Asterisk has a compile time
32-bit unsigned int version which is incremented whenever some major new
bug is fixed. When
Okay fellow Asterisk users... Listen up (or perhaps more appropriately,
read down).
The Asterisk community is growing at a remarkable pace. I know there are
thousands of you out there -- in fact there are over eight *thousand*
subscribers to asterisk-users alone, and almost one *thousand*
I am having same problem and i was never successful in connecting to
digium.com or asterisk.org or asteriskpbx.org for last three days.
We've been moving our office but we are now in the new location and
hopefully won't have any more trouble. Thanks!
Mark
We have just one dual athlon system and we had trouble with the TE405P in
the 32-bit slots of that system. We have not heard of anyone else having
any problems like we had on that one system, but still had to say it i
guess. The failure is that it constantly is taking errors, not the kind
of
no why would it need to do that? If I put one in my 64 bit slots the
machine won't boot.
Digium X100P's do work in 64-bit and 3.3V slots just fine.
Mark
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Currently in progress of trying to debug similar
problem on my own system. Sometimes it happened
during call transfers, but this last time,
it happened all by itself at 4:00 AM, no calls even
close. Complete system Freeze, Nothing at all
workings, except the reset button.
You setup is
I had two system freezes this weekend, first time. I just setup
Musiconhold. The kernel panic referenced mpg123. I turned off
musiconhold until I could look into it more.
Again, please post your comments *on the tracking bug* number 963, at
bugs.digium.com. Please include whether you have
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
Mark
On Sun, 8
This seems to only apply to non-zap channels participating in the
conference, incidently.
On Fri, 6 Feb 2004, mattf wrote:
Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP
kernel and have about 30 channels in conference. Here's the bug listing:
Usually the cards seat pretty well. Do you have a green or blue
TDM40B card?
Mark
On Wed, 4 Feb 2004, Greg Kedrovsky wrote:
I have a TDM40B, 4-port fxs card. Each port seems to have it's own
little board on the fxs card. Each little board is not sodered in, but
rather hangs (I have a
Asterisk 0.7.2 is now released and contains lots and lots of bug fixes
from the bug tracker. Highly recommended for people running 0.7.1.
Mark
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To
Will there be no FXO daughter boards for the TDM400?
There will be. Units are again in production after having an issue that
had us stuck for about 2 months.
Mark
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in the *short term* just add:
noload = chan_iax.so
to your modules.conf
Eventually we will move chan_iax2.c to chan_iax.c and chan_iax.c will
become chan_iax1.c and will likely not be a default part of the build
process.
Mark
On Sat, 31 Jan 2004, Vic Cross wrote:
G'day list,
I am getting
You cannot allocate a static host (host=foo.com) and also use
registration. If you are going to have one register with the other (it is
rarely valuable to have both register with both sides) then you should use
host=dynamic. Alternatively you can not use registration and just use
statics.
Mark
Latest CVS should not detect 'f' except in the case of a real fax.
Mark
On Thu, 29 Jan 2004, Brent Franks wrote:
Christian,
You can change channel.c source code to be more forgiving of
unrecognized DTMF tones.
Look for my addition near the bottom of this struct:
else if (digit ==
You need CVS with echocancel=yes, echotraining=yes,
echocancelwhenbridged=yes. We'll release a 0.7.2 before too long since we
haven't been able to branch 1.0 yet.
Mark
On Thu, 29 Jan 2004, Stephen R. Besch wrote:
Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in
What *I* want to know is why someone has not made a CHEAP PCI card with
4, 8, or 16 of these DSPs on it. This kind of card would provide
hardware assisted DSP functions as well as patent indemnification.
Would you even have to USE the DPSs in order to be patent indemnified?
Using the DSP
The restarts only occur on idle channels. However, what is interesting is
that according to this log, the *switch* requests a restart on channel 20
(which if my calculations are correct, is the channel that has the call on
it). You can see this because we log the messages on getting RESTART_ACK
Asterisk 0.7.1 has been released fixing a few minor bugs. Thanks again to
the bug marshalls, especially Malcolm and bkw.
Mark
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To UNSUBSCRIBE or
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
Mark
p.s. there was no 0.6.0 release.
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I plan on removing chan_iax from the normal build process, and making
chan_iax2 register itself as both IAX and IAX2. IAX1 if built will
register itself as IAX1. CVS asterisk has already been updated such
that IAX1 can be used to identify an IAX channel.
The removal of chan_iax from the normal
Red alarm on an X100P indicates that there is no battery on the phone line
for an extended period of time... You might want to contact Digium tech
support.
Mark
On Thu, 8 Jan 2004, Christian Hoffmeyer wrote:
- Original Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL
No, it's some side effect of the 2.6 changes... strtok goes away in 2.6,
i'll try to fix it.
Mark
On Thu, 8 Jan 2004, TC wrote:
I see http://bugs.digium.com/bug_view_page.php?bug_id=769
that might be your issue
- Original Message -
From: Sean Swallow [EMAIL PROTECTED]
To:
I still think we need something more fine grained. I think we can add the
asterisk-biz list, and eventually something akin to a newbie list, but
need a more appropriate name, IMHO.
Mark
On Fri, 9 Jan 2004, Panny Malialis wrote:
Just a suggestion,
Could Digium make lists.digium.com
Prompted by the recent discussion on the mailing list regarding the
Asterisk development and release process (or lack thereof), John Todd,
Thorsten Lockert, Brian K. West, and myself have put together a plan to
address the most significant two legitimate concerns that have been
expressed regarding
Thanks for the reply. Yup, sure enough it appears the calling party name
is in the facility message. I get the following, where the 'ATLANTA' and
'GA' sections are the calling party name.
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 2 (reference 527/0x20F) (Originator)
the only thing that i can think at this point is that
mark doesn't want sip to work through nat.
Right, you've caught me. My goal has always been to prevent SIP from
actually working in Asterisk, because deep down I really just want the
whole project to fail.
Or maybe it's because (a) I was
Sounds like something nasty being printed. If you run asterisk in the
background (without -vvvgc) and don't attach to it do you hear it still?
Mark
On Fri, 2 Jan 2004, Patrick wrote:
On Fri, 2004-01-02 at 08:31, Florian Overkamp wrote:
Hi,
I've sent this to asterisk-dev recently, but
You can upgrade to revision E/F for free (other than shipping cost). Just
contact Digium sales and tell them you need to swap your TDM cards.
Mark
On Sun, 28 Dec 2003, Victor Rini wrote:
Tilghman,
I have a feeling we're getting somewhere.
I ordered three cards the very day they went on
For anyone that had been planning on meeting:
In order to try to finish the TE405P before the end of the year, I have
had to delay my trip to Paris by three days.
I will now be leaving for Paris on the 21st arriving the 22nd, instead of
the 18th arriving the 19th.
Please reschedule any meetings
I've already got Asterisk using the new recv only method too.
Mark
On Mon, 15 Dec 2003, Christian Stredicke wrote:
Hi folks,
in order to establish backward compatibility we made an image that
automatically detects if the other side does not support RFC3264. Please try
it out, we would be
Just noload chan_iax right?
Mark
On Sun, 14 Dec 2003, Dan wrote:
Hi,
How can I do to register an Asterisk server using just IAX2?
If I have a line like the following in iax.conf
register = user:[EMAIL PROTECTED]
the server tries to register with both IAX and IAX2.
if the line is :
I've been running 0.5.0, which is dated sometime in September of this
year and I've noticed a couple of new features in more recent code that
I'd like to use, but am hesitant to go w/ CVS code. My system is not
exactly a production system, it's mostly test, but I'm still leery of
the fresh
a friend just notified my of the the cover story of freeX 1'2004:
Linux als Telefonanalage (engl.: Linux as PBX)
http://www.cul.de/freex.html
Notice that gnophone was on the cover. We really need to resurect it as a
project!
Mark
___
I suppose trunk groups on SIP would be interesting.
Mark
On Tue, 9 Dec 2003, Nicolas Bougues wrote:
On Tue, Dec 09, 2003 at 08:28:27AM +0100, Florian Overkamp wrote:
Registration cascading is not possible (I think) but could it be solved
with a shared dial route:
Instead of
I've continued to have lockups since last week and this is really beginning
to drive me crazy. Could this be a problem with specific hardware or a
specific setting in the configuration files that causes these lockups for
some and not others? If everyone that is experiencing Asterisk lockups
Should be up now. Found/hopefully fixed a really essoterric iax2
deadlock.
Mark
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Good, glad to hear things are better. Without getting into too much techie
detail, what was the root problem?
There was just race that I introduced a while back. If calls came in
while a reload was taking place in IAX2, bad things would happen. Now
it's fixed. Originally I was thinking it
The FWD bridge is currently down, as I am looking to try to find someone
else to host the IAX2 to SIP gateway so that iaxtel can remain strictly
IAX2.
Mark
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Would have probably been more appropriate to at least announce that
iax was going to disappear at some specific date, as opposed to folks
randomly discoverying it and chasing problems. (Kind of related to why
there isn't a marketing plan.)
Sorry, it was something of a side effect of some
You can definitely do that with GSM and G.729 when running IAX / IAX2.
Mark
On Tue, 2 Dec 2003 [EMAIL PROTECTED] wrote:
I wonder which voice codec they use, they say one can use a
28k modem using their service which rules out ilbc.
On Mon, 1 Dec 2003 17:34:41 -0500
Chris HARIGA [EMAIL
Just to inform the community ... i received an
offer last week for 1 week of asterisk training
+/-2USD !! We can't aford this !
Is that USD 20.000,- as in twenty thousand US dollars, or is have
someone played around with the keyboard? If so - who the fuck can afford
to pay
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 backwards (C) symbol on front
Those things generally happen in Amsterdam. And in Kristiania in
Copenhagen. The usual problem: Smoking too much pot
Actually we just had dinner and had left our things in his car which
(according to the police inspector) was entered through the trunk using a
half a tennis ball.
Mark
Hmmm... what size was that T shirt ? (c;
Large.
Mark
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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 Sun, 30 Nov 2003, zoa wrote:
Count me and one of my collegue's in.
How long are you staying in Paris ? The 19th might be a bit early for us,
but then
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together in Paris sometime near there. Any one out
there interested? Anyone in Paris who could help organize something like
that? :)
Mark
___
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with status or IM. So maybe there is some quasi presence
avaible? I
Assuming you haven't cvs updated yet I can look at this problem but I need
matching sources/binaries/cores. If you've cvs updated, there isn't much
I can do.
Mark
On Thu, 27 Nov 2003, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 27 November 2003 05:57,
Then i started asterisk, it opens the D-Channel and
everything is still ok. I left the system in this state
and it survived one night without problems. But immediately
after the first call (B-Channel) the system memory is overwritten
and bad things happen.
I looked through the wct1xxp.c and
You're forgetting to answer the line first.
Mark
On Fri, 21 Nov 2003, WipeOut wrote:
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
DTMF is used in some places. Japan uses FSK, but a rather different
message format. There isn't a whole lot of global standardisation in CLI!
Not only that but I believe they use different frequencies, and utilize a
parity bit as well.
Mark
___
I think we have to figure out what the difference is. It'll take going
through the mgcp debug output to see what is going on.
Mark
On Mon, 24 Nov 2003, ProvoCityPower wrote:
We are working on a new implementation of asterisk. We are using a fiber-served
WorldWide Packet switch at the home
Again, can you please confirm you are neither running serial console *nor*
graphical console (e.g. framebuffer). If you can call into the office we
can ssh in and take a look at the configuration.
Mark
On Fri, 21 Nov 2003, Scott Stingel wrote:
(Apologies: starting this as a new thread - I'm
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?) would be
Let me clarify my feelings:
I believe the API should look something like this:
struct ast_features {
/* Private data for features, which ones are enabled, state
information, etc */
};
/* Apply var/value pair to the feature set, return 0 on success, -1 if
this isn't a
Yes, it's in. If you cvs update you should see an entry in
zaptel.conf.sample and a brief description.
Mark
On Thu, 20 Nov 2003, Johnson, Randy wrote:
I could swear that I remember seeing some announcement somewhere that Zaptel
now supported drop-and-insert across spans on a TE410P, but now
I am not sure a newbies list would help all that much, all that would
happen is that they would cross post to both lists and we would get
everything twice.. What may be better would be either a better way to
search the list archive or a new users FAQ, of course the FAQ option
requires that
I've made some updates that may help alleviate these problems on TE410P on
E1. I'd like you to test and tell me if that helps. Also, I'd like you
to confirm that you are using neither serial console nor frame buffer
(graphical) console.
Mark
On Thu, 20 Nov 2003, Scott Stingel wrote:
Hi all-
show channels and zap show channel foo should be helpful. Find me
on IRC if worst comes to worst.
Mark
On Wed, 19 Nov 2003, Matt Lawson wrote:
I recently updated (fresh checkout) to the newest zaptel and Asterisk.
The one I was using before was a couple of months old.
After updating, my
That's a question for Greg.
Mark
On Thu, 20 Nov 2003, Isamar Maia wrote:
Any prevision or estimative of final price ? :-)
Isamar
On Wed, 19 Nov 2003, Mark Spencer wrote:
We *are* making progress, and i have a running prototype, however the
production board is having some trouble
Why don't we just add it on the DIgium list server, wouldn't that make
more sense, to have a single place for all list memberships?
Mark
On Wed, 19 Nov 2003, Asterisk online forums wrote:
Adam,
We had discussions about business list or business implementation for
really long time, now it
You now get ${SIPDOMAIN} when an incoming call comes in via SIP, so for
now you could do something like:
exten = _X.,1,Goto(cxt-${SIPDOMAIN},s,1)
and then have:
[cxt-digium.com]
exten =
and so on.
Mark
On Tue, 18 Nov 2003, Tristan 'Minty' Colgate wrote:
Hi,
Is it possible to pick
yes, it's provided as an IE during registration.
Mark
On Mon, 17 Nov 2003, Dan wrote:
Hi all,
There is any MWI implementation in IAX2 to be used by a software IAX2 phone?
Thanks,
Dan
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If I am dialing with a Bt101 or something that sends all the digits in a
single packet, it works great. It fails miserably, however, if I'm dialing
from a phone on an FXS port, or if I'm trying to do this on an answered
call.
Zap devices should handle this fine (maybe even MGCP), but SIP
you can do it with immediate=yes in zapata.conf on your FXO signalled
(FXS) interfaces, and then dump them in a context with s extension that
dials another phone.
extensions.conf:
[autodialz1]
exten = s,1,Dial(Zap/1)
exten = s,2,Congestion
[autodialz2]
exten = s,1,Dial(Zap/2)
exten =
it's implemented on the zap side (which is now configurable with
jitterbuffers=foo in zapata.conf.
Mark
On Wed, 12 Nov 2003, Matteo Brancaleoni wrote:
mmmh... I'm not sure ig chan_sip has jitter buffer.
I think that there isn't a jb in sip,
but correct me if I'm wrong.
Matteo.
Il lun,
I'll try to call them tonight.
Mark
On Wed, 12 Nov 2003, costas wrote:
I guess people are pissed off with them and are looking at the alternatives. I think
they are charging too much money for it. Also they must compete against MS free
Personal Server (SQL Server but not optimized) and
If you're looking for encryption enough to foil casual sniffers why not just
use something like DES1 or even straightforward and fast XOR encryption and
use a hash of the call ID, trunking messages and rekey with new values
every few seconds (let's say with a random sample of the unencrypted
So if you only have smp systems with ohci and no zaptel cards (because it's a
sip/iax2 gateway) then you're screwed?
You can always get an x100p...
mark
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Facility IE seems to be the ioctl of the ISDN world. Even if you look
at the intel app note, it leaves out a lot of details about what's going
on. I'd love to get TBCT if anyone can find me a good, complete spec for
implementing it.
Mark
On Mon, 10 Nov 2003, Gene Kochanowsky wrote:
Hi Gavin,
The OpenOffice.org project have a marketing subproject that has been
doing a lot of non-development things. Coordinating pressreleases,
arranging conferences, participation in other conferences, training
material etc. Maybe we should look into stealing ideas from that project?
Since our
Please confirm this is fixed now in CVS.
Mark
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Asterisk has got to be about the best kept secret in telephony. I've seen
numerous articles on slashdot about VoIP, even in relation to Linux and
only *once* has the post even mentioned Asterisk. Am I missing something,
or is Asterisk clearly a good potential player in any kind of linux-based
There would have to be a corresponding change in the SIP dialog or in the
actual audio sent both ways. Can you provide some information on how it
has changed?
Mark
On Fri, 7 Nov 2003 [EMAIL PROTECTED] wrote:
Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the
I think there are issues with combining flash-hook supervised transfers
with meetme conference bridges. Can you find out if that took place, i.e.
someone tries to transfer into a meetme conference?
Mark
On Thu, 6 Nov 2003, Andy Hester wrote:
Wondered if anybody might have some ideas about
No, I have Meetme on an extension, but the users don't even know about it
and the ext# for it is in a completely different range.
Hrm okay.
I don't know if I explained it adequately looking back on my post. The
situation persisted over 8 to 10 hours and through numerous calls. what
seems
You can change it in digits.h in zaptel, recompile, reload all drivers or
reboot.
Mark
On Wed, 5 Nov 2003, Mark Hagler wrote:
Is there a way to get my X100P card to dial slower on the line? Mine
seems to dial the digits too short/fast for the switch to catch all of the
digits and roughly
I am more interested to support S100U from Digium, if I can get a Windows
driver for it.
If you are serious about doing this, and you are willing to LGPL or GPL
your final product (remember, even GPL'ing your code, you can still
licenses it commercial as we are able to do with Asterisk), I can
- Should I implement IAX or IAX2? What's the main difference, other
than IAX2 supporting trunking (which according to the docs needs a
Zaptel timing source).
IAX2 without any question. You will not be required to run trunk mode in
your case, especially if you're just doing it
I don't plan on using it. I will use mine, which is created in wxWindows
and C++ and will run on Winsucks, UN*X and Mac.
Yes, someday it will get released, maybe even the code if people are nice.
I think it's just a race to see who makes the first GPL'd or LGPL'd IAX
client for Windows (or
As the library is under LGPL (is not true?), I intend to keep this
application as a freeware only...
I want to add new features, but for one of them I need new functions
implemented in the library (like multiple codecs support, message waiting
indicator, conferencing, etc.).
There is no
This was triggered by the lack of an mp3 encoder. Without a backtrace
there's no way to know it's fixed for sure, but if you cvs update it
should at least fail cleanly and if not please place a bug in the bug
tracker.
Mark
On Tue, 28 Oct 2003, Alexandru Coseru wrote:
== Parsing
Update your zaptel.
On Tue, 28 Oct 2003, Bartosz Jozwiak wrote:
Today's CVS gives me an error while compiling:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1076: `ZT_ECHOTRAIN' undeclared (first use in this function)
chan_zap.c:1076: (Each undeclared identifier is reported only once
Just as a heads up, soon, I will be merging Thorston Lockhart's new tagged
CVS archive over to Digium. This will mean you have to do a *clean* check
out of asterisk, zaptel, libpri, etc.
For those of you with *localized changes*, please be sure to do:
# cvs diff -u ../my-asterisk-changes.diff
1 - 100 of 386 matches
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