That's not something that is likely to be supported. Any configure
script in the tree will be run via the top-level build process, as
needed. Is there some reason you think you need to run the other
configure scripts yourself?
On 03/05/2014 08:54 AM, Gianluca Merlo wrote:
Hello everyone,
I
The packages currently do not support SRTP.
On 06/03/2013 10:56 AM, Daniel Pocock wrote:
I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org
I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
The SRTP support appears to be missing
On 06/03/2013 12:03 PM, Daniel Pocock wrote:
I tried building manually from the source RPM
Before running rpmbuild, I installed libsrtp-devel and I notice that
res_srtp.so is generated during the build
However, the rpmbuild fails for other reasons (see the other email I
sent to the list
On 05/21/2013 10:19 AM, Ahmed Munir wrote:
Hi,
Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily
basis which was working perfect. Now in couple of months back, the
logrotate feature is not working at all but simply appending the logs
in 'messages' file. Listing down down
On 05/07/2013 05:13 AM, Olivier wrote:
2013/5/7 Matthew Jordan mjor...@digium.com mailto:mjor...@digium.com
2. It appears as if you're running a modified version of Asterisk, in
which case all bets are off. This works fine on the Linux build
agents,
which is what we use to
On 01/03/2013 02:23 PM, Markus Weiler wrote:
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
I think he means 10007.
--
On 10/03/2012 10:46 AM, Eric Wieling wrote:
A port is not a door if there is nothing listening on the port.
Open ports are not a security issue. Stuff running on open ports are.
Do you have some external software listening on those ports when there isn't an
active call? Asterisk isn't
On 08/28/2012 10:04 AM, Andrew Latham wrote:
On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote:
2012-08-28 16:44, Andrew Latham skrev:
Try this to test with
http://www.digium.com/en/products/ivr/audio-converter.php and compare
your output first...
Interesting. Didn't
On 08/28/2012 10:32 AM, Danny Nicholas wrote:
Does the .c program compile stand-alone or as an add-on?
g++ check_sounds.c
check_sounds.c: In function âint main(int, char**)â:
check_sounds.c:152: error: invalid conversion from âvoid*â to âdirent**â
check_sounds.c:154: error: invalid conversion
On 06/12/2012 02:56 PM, Danny Dias wrote:
Hi,
I'm just trying to install the DPMA on my Asterisk. I already made the upgrade
from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did:
/mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
/
*compiling Asterisk-Cert2
On 06/05/2012 10:23 AM, Chet W. Stevens wrote:
During testing with the Digium phones I have run into a problem where I try to
make a change to the sip device name. I make the device name change in
sip.conf
then make the matching change to the lines in res_digium_phone.conf. I then do
'sip
On 05/22/2012 04:54 PM, Danny Dias wrote:
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
msg.WAV
That is perfectly normal. The .txt file is metadata that contains things like
caller ID and duration. Asterisk will also save voicemails into every format
you
On 04/27/2012 01:39 PM, Don Kelly wrote:
What flavor are flashphoner minties?
--Don
Dailing flavored. What else?
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On 03/06/2012 12:31 PM, Ron Bergin wrote:
Mathew,
Each of those odbc modules are unavailable i.e., marked with XXX
I even deleted the asterisk build directory and started over, but had the
same results.
What prereqs do I need besides these:
mysql.i386
On 03/06/2012 03:44 PM, Karl Fife wrote:
It's not a question of whether the default directory permissions are
appropriate. I agree with those.
What we're talking about here is what happens during updates to an existing
directory. I can't see any rationale for changing the group permissions.
On 03/06/2012 04:24 PM, Patrick Lists wrote:
On 06-03-12 23:07, Karl Fife wrote:
Yep. That's what's happening. I'll file a bug.
AFAICT it's not a bug but the way RPM works.
Regards,
Patrick
He didn't suggest that he was talking about RPMs. If that's the case, then I
take back
On 03/05/2012 06:34 AM, Eric Germann wrote:
Does anyone have an idea on when 1.8.9.3 might show up in the RPM
repositories?
Thanks!
EKG
They should be available now.
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On 03/05/2012 01:49 PM, Eric Germann wrote:
Will a 1.8.10.0 build be imminent or should we go ahead and push this in to
production with testing?
Thanks!
EKG
~20 minutes
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On 03/05/2012 06:00 PM, Lefteris Zafiris wrote:
Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled
against 1.8.7:
asterisk18-addons-core-1.8.7.0-2_centos5
asterisk18-addons-mysql-1.8.7.0-2_centos5
Is this a problem with the repo? Are these packages
obsolete/unmaintained
On 03/05/2012 06:22 PM, Karl Fife wrote:
I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably
earlier versions too) remove the group write permissions from
/etc/asterisk/. which is different than 1.4. And 1.6.
Is this expected behavior?
If so, what's the rationale?
If not,
On 02/26/2012 06:22 PM, Patrick Lists wrote:
On 25-02-12 19:47, Jason Parker wrote:
yum and rpm do not support downgrades.
Incorrect. There is yum downgrade. See man yum.
yum downgrade is extremely broken. It fails, often, potentially leaving a
system in an unrecoverable state
version when
using
the repository? For example, Asterisk 1.8.9.2 is available now. But I want to
use 1.8.9.1. Can I downgrade somehow? I want to test NAT bug issue.
Thanks
On Thu, Feb 23, 2012 at 11:15 AM, Jason Parker jpar...@digium.com
mailto:jpar...@digium.com wrote:
On 02/23/2012
On 02/23/2012 10:09 AM, Ast Coder wrote:
Hi,
I have followed instruction
on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites
to
add Digium Asterisk repositories but doing a, yum search asterisk only shows
me Asterisk 1.4, 1.6, and 1.8. There is
On 02/21/2012 02:55 PM, Bryant Zimmerman wrote:
Ok I now have the basics for dynamic parking working but for some reason when
a
caller calls in and is parked with a transfer the return call dials the sip
peer
of the caller and not hte peer of the last party that parked the call. Anyone
On 02/21/2012 05:34 PM, Stephen Brown wrote:
application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000
/var/lib/asterisk/sounds/music/Rolling In The Deep.mp3
Probably unrelated to your issue, but you're going to want to quote that
filename.
--
On 01/30/2012 11:06 AM, Eric Germann wrote:
We mirror off http://packages.asterisk.org to a staging server, then update
from there.
When will this show up on packages.asterisk.org?
Thanks!
EKG
The RPMs should be there in a few minutes.
--
On 12/28/2011 03:10 PM, Danny Nicholas wrote:
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains
new
On 12/12/2011 09:26 AM, Danny Nicholas wrote:
I'm wondering if the bind 161 as root statement is a mis-statement or
if not, maybe somebody like Tzafir can explain why since none of the
other Asterisk binds require root access (this message is still in
10.0-rc3).
This is accurate. Non-root
On 11/15/2011 09:58 AM, Tony Mountifield wrote:
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74.
My two questions are:
1. Is there a list of these standard assignments somewhere? Googling did
On 11/15/2011 10:42 AM, Tony Mountifield wrote:
Yes, I was hoping to use such a system user and group for asterisk, which
would not conflict with any other system package I might install in the
future, by virtue of being reserved for asterisk.
There shouldn't be any conflict either way.
On 10/20/2011 05:16 PM, Paul Belanger wrote:
Greetings,
If you are planning on attending Astricon, please take the time to
attend the GPG key signing event. More information can be found on
the wiki page[1].
[1]
On 10/18/2011 09:52 PM, Luke Hamburg wrote:
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
It is indeed a bug, but it's not the bug you referenced. This issue
only exists in 1.8.8.0-rc1. It has been fixed for 1.8.8.0-rc2 which
will be released
On 10/17/2011 02:22 PM, Ioan Indreias wrote:
Hello,
Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM
version from Asterisk repo I found that asterisknow-version is needed
by package asterisk18-core-1.8.7.0-2
How could this be explained?
Best regards,
Ioan
The
On 09/30/2011 09:53 AM, Tony Mountifield wrote:
In article 4e85d19f.4090...@digium.com,
Kevin P. Fleming kpflem...@digium.com wrote:
This is why the output was changed to microseconds from milliseconds; in
the older version, the lowest number that should be shown was 1
millisecond, even if
On 07/19/2011 01:02 PM, Michael wrote:
On Tue, Jul 19, 2011 at 3:34 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
You don't need to install asterisk-addons to be able to store CDRs; you need
them to be able to store CDRs in MySQL specifically. If you
On 05/17/2011 07:18 AM, Stefan Gofferje wrote:
On 04/17/2011 02:13 AM, Stefan Gofferje wrote:
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
I finally figured it out.
For facebook chat to work you have to
On 05/16/2011 08:36 AM, Jerry Geis wrote:
I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d
directory.
[digium-current]
name=CentOS-$releasever - Digium - Current
baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
On 05/12/2011 02:46 PM, Jason Parker wrote:
I'll make it a point to respond to this email when the new builds are available.
These builds are now available.
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On 05/12/2011 02:40 PM, Cassius Smith wrote:
Hi all
Usually I build asterisk from source, but recently have been doing a
couple of test installations with packages from the Digium repository.
About how long does it take to get from new release announcement into the
Digium RPM repository?
On 05/06/2011 01:30 PM, Bob Beers wrote:
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if %{paevar} == PAE
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
You'd probably want to modify the kmodtool that comes with it, to
On 03/15/2011 12:34 PM, Fellipe Paes wrote:
why I can't use _. in my dialplan?
Because it matches everything. In this case, it's matching the 'h' exten. So
when the call gets hung up, it goes to _. and does what you're seeing.
--
On 02/23/2011 12:43 PM, vip killa wrote:
I recognize all the options given yet as I explained before they are not viable.
I do not have the resources to pay someone, I do not have the expertise to fix
this issue because according to an asterisk developer any fix in that area
would be deeply
On 02/02/2011 02:14 PM, Frank Liu wrote:
Hi there,
Per the instruction from http://www.asterisk.org/downloads/yum , I
setup the yum repository on my Centos 5 x86_64 machine and did a
yum install asterisk18 asterisk18-configs
then I startup the asterisk (with no changes to config) just to see
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to
attempt to kill a thread is rarely successful.
On 01/19/2011 04:41 AM, Ishfaq Malik wrote:
Hi
Does anyone have any idea how long it will take for the new release of
asterisk 1.8 to make it to the digium yum repositories?
Thanks
Ish
They've been there since yesterday afternoon. It's possible that you hit the
repository before the
On 12/20/2010 11:35 AM, Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2.
On 12/02/2010 02:03 PM, Danny Nicholas wrote:
Hi gang,
We are moving our computers from a cluster of physical machines to a VMWARE
server with virtual machines. We investigated and are looking to replace our
TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers
from
On 07/28/2010 11:32 AM, Tilghman Lesher wrote:
They permit what packets will even reach user2
It should also be pointed out that the config option is permit, and not
allow.
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On 07/19/2010 01:23 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
Sent: Monday, July 19, 2010 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice
On 07/15/2010 08:16 AM, Vasiliy G Tolstov wrote:
Hello.
Who can add asterisk16-xmpp module to packages.asterisk.org or build
asterisk with support xmpp and update packages?
Thank You.
This is something we've been considering for a while. It should make its way
onto the list shortly.
--
On 05/26/2010 08:00 PM, cov...@ccs.covici.com wrote:
From another thread, I blacklisted netjet and now things are working.
But I wonder what is going on here and where did netjet come from -- it
doesn't look like an dahdi module to me.
It comes from mISDN. It is a very badly misbehaving
On 05/12/2010 01:03 PM, Robert Wagner wrote:
Hi,
when i include a sip configuration from another file in my sip.conf
using #include /etc/asterisk/sip-sipgate.conf everything seems to be
working.
The peer is listed when i execute sip show peers and Status is OK.
But the peer is not listed
Michael Nausch wrote:
HI,
I tried to install asterisk and mISDN via
http://www.asterisk.org/downloads/yum
My machine is running with kernel-2.6.18-164.15.1.el5.i686
Packages for that kernel version were missing. That was an oversight and has
been corrected. A `yum update` should be
Olivier wrote:
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such
a way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
Pablo Ruiz wrote:
Hello,
Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary
packages at packages.asterisk.org http://packages.asterisk.org?
Greets.
Packages for 1.6.2 will be available Real Soon Now. It's near the top of my
short list.
They exist, and are sitting
bruce bruce wrote:
Thanks for the update Jason,
How do the upgrades work if v1.6.0 is already install and one wants to
upgrade to 1.6.2 (once it's available)?
yum upgrade asterisk*
???
Thanks
It should be as easy as a `yum update`. That's the goal, anyways.
--
Brian J. Murrell wrote:
I wonder if Asterisk's skinny/sccp channel driver could be used as a
client to register with a Cisco PBX. That is, along with a SIP
client, say, have Asterisk and said SIP client stand in for a Cisco
phone, or an IP Communicator.
Anyone done this?
Cheers,
b.
Jay Vocaire wrote:
Thanks for researching this for me. If you actually look at the link
you sent me, you will see that the latest is:
asterisk16-core-1.6.0.21-1_centos5.x86_64.rpm 20-Jan-2010 15:45 11M
So, we come back to my original question: is there a reason for the
delay on getting
stephen.hindma...@bt.com wrote:
rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec
snip
error: Failed build dependencies:
kernel-devel = 2.6.18-164.11.1.el5 is needed by
dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386
Add a --target=i686 to your rpmbuild
Brian wrote:
Each time the server is rebooted Asterisk duly
deletes the manually created /var/run/asterisk directory - quite why it
does this I just don't know - perhaps it is a bug?
Your assumption is incorrect. Some Linux distributions will empty /var/run/ on
boot, just as they do with
Doug Lytle wrote:
Dave Fullerton wrote:
Note num and not number I don't know if that was a change from 1.4
to 1.6 or if Doug mistyped it.
Not a mistype. I've been using number all along, but looking at the
docs shows that I've been incorrect. It must concatenate the number
down to
Noah I. Engelberth wrote:
I’ve been spending the day trying to get IMAP_STORAGE on my test box, to
evaluate for production, but I’m having no luck getting uw-imap to
build. I’ve tried installing it from an upstream package, but Asterisk
still isn’t finding it to compile –with-imap. My google
Mark Hulber wrote:
It looks like there's a problem with the location or naming of the Extra
SLN16 sounds:
This has already been fixed in the 1.6.1 branch. It should make its way into
the next releases.
See 1.6.1 revision 212386.
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D Tucny wrote:
%changelog
[snip]
awesomeness here
[/snip]
I'm speechless. This is far beyond what I could have possibly hoped for. It is
also extremely accurate.
Thank you very much for this. I'll be sure to keep this (and others) up to date
in the future.
D Tucny wrote:
2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com
Hi, Axel.
Axel Thimm wrote:
How about merging in your changes/improvements/new packages with
ATrpms (and automatically later into rpmrepo.org
http://rpmrepo.org)? That way we
Robert Broyles wrote:
I saw some of the heat about the $20 bounty earlier. So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)
I'm in need of getting this bug fixed. Bug has all of the details, but
basically 1.4.22 broke
Tilghman Lesher wrote:
On Wednesday 04 March 2009 10:24:16 Robert Broyles wrote:
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
Murf is plenty legal; he simply doesn't consume
Jason Lixfeld wrote:
This link
(http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/
) seems to indicate that in order to upgrade AsteriskNOW v1.5 from
Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package.
Does anyone know where to find that
This should now be fixed. If you want to force an update, you can do something
like `yum clean metadata; yum update`
Jason Parker wrote:
It apparently isn't built with IMAP support. That would be a bug in my
packaging. I'll see what I can do with it.
Jason Lixfeld wrote:
I'm having some
It apparently isn't built with IMAP support. That would be a bug in my
packaging. I'll see what I can do with it.
Jason Lixfeld wrote:
I'm having some issues getting app_voicemail_imapstorage to talk to my
IMAP server. From imapstorage.txt, I've got the voicemail.conf
configured
Jerry Geis wrote:
wct4xxp: sh: /sbin/ztcfg: No such file or directory
FATAL: Error running install command for wct4xxp
[FAILED]
Hmm.. Something in /etc/modprobe.conf, /etc/modules.conf, or
/etc/modprobe.d/?
Jason Parker wrote:
I just wanted to post this so that it was out there and Googleable. Hopefully
it will save other people a bit of time.
If you have a Cisco phone (I was testing with a 7970, though presumably it
would
affect 7960 and others as well) that is looping trying to fetch
I just wanted to post this so that it was out there and Googleable. Hopefully
it will save other people a bit of time.
If you have a Cisco phone (I was testing with a 7970, though presumably it would
affect 7960 and others as well) that is looping trying to fetch the CTL tlv file
- it may be
Philipp Kempgen wrote:
I would suggest screen ( http://en.wikipedia.org/wiki/GNU_Screen ).
screen doesn't solve the security aspect of your question though.
Grüße,
Philipp Kempgen
Actually, it could. What I've done before, is give out an unprivileged account
on the box (or some
Steve Totaro wrote:
This looks like it may be your problem.
http://bugs.digium.com/view.php?id=9592
(0070069)
qwell - administrator
09-06-07 17:05
Closing.
The simple solution here is to just comment out the #define USE_RTC in
ztdummy.c. The ztxen module does not appear to be
מוישי ברעוודה wrote:
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
Brent Davidson wrote:
Do they mean 1.4.20 instead of 1.4.10? If not, then this message was
seriously delayed :-D
-Brent
Zaptel, not Asterisk. :)
1.4.10 is correct.
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Continuing the top-posting madness...
For future reference (and for the archives), you could have done `make
dist-clean` and re-run configure, rather than remove the directory.
Kyle Gibbons wrote:
All,
Thank you very much for your help, I have solved the problem. After
installing
Joshua Kinard wrote:
-Original Message-
You probably mean a T100P? The single E1/T1 card? Been a few years but I
remember seeing the NMI Errors on a HP DL380 (the Intel dual Xeon
model).
Nah, it's classified as a D110P, although the driver says TE110P. And I
checked to make sure
with this card? Anyone know if there are plans for
a PCI-e analog card for FXO use?
Digium already makes PCI Express analog cards - AEX800 and AEX2400.
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Digium
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Dirk Enrique Seiffert wrote:
I guess this
libtool-ltdl-1.5.22-6.1
... which is installed.
Thanks
Enrique
I believe you're looking for libtool-ltdl-dev(el)
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v1.2.9.1-BRIstuffed-0.3.0-PRE-1q
SCCP firmware
Load File: TERM70.7-0-1-0s
App Load ID: Jar70.2-9-0-117.sbn
JVM Load ID: CVM70.2-0-0-112.sbn
OS Load ID: cnu70.2-7-4-134.sbn
Boot Load ID: 7970_64060118.bin
--
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Digium
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Philip Prindeville wrote:
[...] There were earlier
experimental versions of IP, but v4 got it right.
and v6 will get it even more right. ;)
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Digium
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asterisk
any bug reports or other info with Google.
This is already fixed in 1.4.15.
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Digium
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http
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Robert McNaught wrote:
...
Anyone know the secret to the dependencies?
Robert McNaught
It's case sensitive. I believe RH uses unixODBC as the package name. You
also need the development package of that.
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/listinfo/asterisk-users
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the patch.
If you have the updated patch with the changes he said were needed, please do
reopen the bug.
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Digium
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To UNSUBSCRIBE or update
it both a revenue source,
and as complicated as possible.
The way I understand it, that $15 doesn't actually even give you the right to
use the SIP firmware. It only gives you the right to access the download
area.
The whole model is silly, at best.
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Jason Parker
Digium
Mobile Switching Centers.
Pretty interesting stuff.
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Jason Parker
Digium
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are trying to record the call in ulaw, or
trying to playback prompts that aren't available in g729.
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more
clear I can make this.
Yes, it was a problem in 1.4.11. However, this has ALREADY been fixed in svn.
It will be in the next release.
If you would like to have this fix, you can run the latest version of svn
branch 1.4.
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Jason Parker
Digium
Bart
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internal molex connectors available, there is another option. Digium
has created an externally powered supply that can be used with these cards.
http://www.digium.com/en/products/hardware/analogpwr.php
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Jason Parker
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, so no amount of upgrading is going to help
with that.
In my opinion (and I think Dan and several others would agree), chan_skinny is
far more stable (and active...) than chan_sccp.
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.
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it... (it was developed against 1.4, so the diff from trunk is
probably trivial)
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]: *** [chan_mobile.o] Erreur 1
make[1]: Leaving directory `/usr/src/asterisk-addons'
Does anyone know what's the problem?
You're trying to use a module written for trunk on 1.4.
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Administrator TOOTAI wrote:
Jason Parker a écrit :
Administrator TOOTAI wrote:
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c: In function `mbl_load_config':
chan_mobile.c:1745: erreur: trop d'arguments
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