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haven't heard any in months.
Step 1 would be to upgrade. You're 13 versions behind.
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to at least one person.
(Note, this code has since been removed, since clearly the date has long
passed.)
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of the suggestions
we got, but we couldn't use it because it was already in use. The original
name (which was never committed) was chan_cellphone.
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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
it... (it was developed against 1.4, so the diff from trunk is
probably trivial)
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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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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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Bart
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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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are trying to record the call in ulaw, or
trying to playback prompts that aren't available in g729.
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Mobile Switching Centers.
Pretty interesting stuff.
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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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the patch.
If you have the updated patch with the changes he said were needed, please do
reopen the bug.
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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
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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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)} :
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
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
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
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
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/?
it was introduced in Zaptel 1.2.17.1. From the description
of zaptel 1.2.17.1 posted to www.asterisk.org:
Added the ability to monitor pre-echo cancellation audio with
ztmonitor
- Noah
Yes, you are correct.
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with a bunch of features.
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need to install svn trunk
(http://svn.digium.com/svn/asterisk-addons/trunk/) if you want to use
chan_mobile.
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has support for speeddials/hints though.
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of Asterisk.
Is anyone able to confirm the same behavior in newer versions? Is there a way
for Asterisk voicemail to behave like regular voicemail where a message remains
New until the caller does something to it (other than simply listening to it)
?
Thanks.
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of the questions/answers for one version are quite relevant to the other.
This fracturing of the community would be very silly in my opinion, and is
extremely unlikely to happen.
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, there are only like 2-3 places where it's
referenced.
It should be immediately obvious how it works.
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- Derek Whitten [EMAIL PROTECTED] wrote:
if i remember right, most of the buttons on those and the 12SP+ phones
don't really work
because there isn't a button template in *
There is a button template, the problem is that most of the softkeys simply
aren't implemented.
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- Chris Nighswonger [EMAIL PROTECTED] wrote:
On 3/29/07, Jason Parker [EMAIL PROTECTED] wrote:
It should be immediately obvious how it works.
Maybe to some who have been in on the skinny/cisco conversation for
awhile. I am not new to c or c++, but am to * and cisco ip phones
.. There is code there that is #ifdef'd out, because it
(mostly) does not work.
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- [EMAIL PROTECTED] wrote:
[snip]
I have a feeling I'm forgetting something fairly easy and stupid, but
I
can't seem to see what it is. Anyone have any suggestions?
Dial(SCCP/[EMAIL PROTECTED])
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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
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
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
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
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
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.
- Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Jason Parker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 22, 2006 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Extensions -- Comments
/listinfo/asterisk-users
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, to avoid
potential issues.
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Take a look at the new sample configs for (random example) SIP, and look at the
new moh options.
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The Digium G.729 binary codec modules (both 32-bit and 64-bit), and register
tool have been updated for use with the Asterisk 1.4 beta on Solaris.
As always, they can be downloaded from
http://ftp.digium.com/pub/telephony/asterisk/g729/
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If you're gonna top post...so am I.
I think you misunderstand what qualify is/does. It appears that you believe
that qualify=1000 means that it'll send out a qualify packet every 1000ms.
This isn't an unreasonable assumption, but it is wrong. The qualify=1000 means
that Asterisk will wait
and other code that uses the API, to fairly easily figure
out how it works. :) If the documentation was so horrible that nobody could
figure it out, there would be no programs that use it - and we both know that
to be false.
Doug.
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.
As for the format 0, I don't know how it's getting that far with no codecs
(though it may in some strange way be related to the typeid). Skinny only
supports ulaw currently (yeah, I know, I'll fix that eventually).
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I cheated, and just added a comments field to the table. Asterisk only reads
fields by name, so extra columns don't hurt at all.
That is, iirc..
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with Asterisk and calling the agents when
a call is placed in a queue is a requisite (I've also read it can be done with
the dialplan, but this app eases the work).
[1] - http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin
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-2.6.16.13-4-obj/i386/smp
make[1]: *** [linux26] Error 2
make[1]: Leaving directory /usr/src/asterisk//zaptel/zaptel-1.4.0-beta2
make: *** [all] Error 2
Thanks
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to the list and would like to know how to search it so that
I do not post any questions that have already been answered (like this one)
- Mark
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- John French [EMAIL PROTECTED] wrote:
I have CentOS 4.4 x86_64 running on an Pentium D 830 dual core
processor
with the smp kernel. Does Asterisk need to be compiled in any special
way to gain performance benefits from this setup?
nope
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Josué
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with SF.net.
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Why not just post the text of the AGI to the wiki page?
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a timing source, such as meetme or iax trunking.
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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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any bug reports or other info with Google.
This is already fixed in 1.4.15.
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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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asterisk
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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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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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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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
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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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
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
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
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
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
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.
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.
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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
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
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 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.
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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 07/28/2010 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 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 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 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
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