On Sat, Jul 31, 2004 at 04:36:51PM -0400, Norman Tomlnis wrote:
I am trying to get PrepaidCID working and, it show's it connecting to the
database correctly. I call the extension and it Asterisk does a core dump.
Can anyone help me?
If you'd like to read over
On Fri, Jul 30, 2004 at 06:33:14PM -0700, [EMAIL PROTECTED] wrote:
Hi,
Has anyone had any success? After a clean install of OpenBSD, I do the following:-
pkg_add ftp://rt.fm/pub/OpenBSD/3.5/packages/i386/gmake-3.80.tgz
pkg_add ftp://rt.fm/pub/OpenBSD/3.5/packages/i386/bison-1.35p1.tgz
Sounds like you'd need to delete res_parking.so since that was replaced with
res_features.so if I've followed the mailing list correctly..
- andrewg
On Mon, Jul 19, 2004 at 05:53:08PM -0700, Nathan Martinez wrote:
Hello,
I was running a very simple test setup with * HEAD 7/15/2004 on Fedora
Ok, just removed this last module works, asterisk is starting without
errors anymore, but I wanted to use ILBC codec so it's importan for me.
Can anyone help me, getting this to work?
Start off with running ulimit -c unlimited before you start asterisk. Once it
crashes, type gdb
on it.
At some stage I was thinking about hooking liarliar up to asterisk to see if
the concept would work. A friend of mine raised the point that some codecs
won't give you the info you're after most likely over IP, so I never looked
further into it.
- andrewg
On Tue, Jul 13, 2004 at 01:24:18PM +0200, Andreas 'TheChaos' Groll wrote:
[EMAIL PROTECTED] wrote:
Start off with running ulimit -c unlimited before you start asterisk. Once
it crashes, type gdb /path/to/asterisk core
From there, enter the following:
bt
x/5i $eip
info registers
Hmm, block is allocated near the top of the stack.
Ack, I don't like the iLBC code for the quick 3 minutes or so I looked at it, but it
wouldn't surprise me if it was overwriting more than it should be on the stack.
Well, I'll hand this off to the developers / people who want to spend longer
I have no idea who Randy Bush is but I found it funny the first article
I found on him was a presentation on why NAT is evil espically for
voice. Now he asserts that NAT traversal is not needed.
http://www.apnic.net/meetings/17/docs/sigs/policy/addrpol-pres-randy-nats.pdf
I read his
On Thu, Jun 17, 2004 at 05:28:00PM +0400, AK wrote:
Hello, ereyone!
I have just installed Asterix on my FreeBSD (-current) box
I'm planning to use it as H323 PBX for softphones
Currently I'm stuck in transfering a call to another machine
running H323 client
When I define forwarding
On Sat, Jun 05, 2004 at 02:35:23PM +1000, dkwok wrote:
I have just compiled the latest cvs 040605 and have this illegal instruction error
when launched asterisk. It is compiled on Via c5 processor. In the asterisk/Makefile
I have set PROC=i586 but it does not help the situation.
Any
iaxclient.sourceforge.net perhaps?
On Tue, May 25, 2004 at 11:07:12AM +0200, [EMAIL PROTECTED] wrote:
Hi There,
i think all VOIP clients for Linux are unusable!
i got testet:
Linphone + Linphonec all in version 12.2
Kphone
gophone
and other...
the only programm that is usable
On Sat, May 22, 2004 at 12:29:11PM +1000, Christopher Lee wrote:
Hi,
G'day,
I've been trying to get the G.729a beta codec running with my remote
Asterisk box that talks IAX2 to my local Asterisk box.
Digium fixed the problem I was having in registering the beta codec, so that
On Sat, May 22, 2004 at 01:05:23PM +1000, Christopher Lee wrote:
Hi Andrew,
Here's the results:-
Okay, doesn't mean so much to me, but it might help someone.
[snipped out most of the above symbols messages]
Reading symbols from /usr/lib/asterisk/modules/format_g726.so...done.
Loaded
On Sat, May 22, 2004 at 03:07:14PM +1000, Christopher Lee wrote:
Okay, doesn't mean so much to me, but it might help someone.
Nor I. I've posted the results to the RT ticket I've already got open with
Digium support in case it helps them as well.
(gdb) x/5i $eip
0x4044e862
the extension names (not that that really means /too/ much), such as .txt or
.conf, as this would increase your 'security' more.
- andrewg
On Mon, Apr 05, 2004 at 04:35:26AM +1200, Matt Riddell wrote:
Can't we get the mailing list to not allow mails with .pif files attached?
Matt Riddell
?
Unfortunately, I don't see this happening any time soon... oh I know, lets make the
mail server check for that happening and block it, so it doesn't screw up peoples
threading :D
- andrewg
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On Mon, Mar 29, 2004 at 12:03:52PM +0200, radan wrote:
Hi all !
Everything works fine but
when I try to make connference,
I receive unable to open channel
I don't have this problem when
I run asterisk as root.
Probably something wiht file permission. Which ?
/dev/zap something, don't
On Tue, Mar 23, 2004 at 07:53:46PM -0600, [EMAIL PROTECTED] wrote:
Hello,
I am interested in knowing if someone has done any work on
IPSec
I've used IPSec on transcontiental links with IAX no problems.
for Asterisk boxes. If so, it will be nice if we can all share our
experiences here.
On Wed, Mar 24, 2004 at 07:09:43AM -0700, Jason Becker wrote:
[EMAIL PROTECTED] wrote:
Another topic of interest is securing the box itself. Does a firewall
(hardware outside of the box or a linux based firewall) suffice the need?
Depends what you are protecting against. If you want
On Wed, Mar 24, 2004 at 08:54:44AM -0800, Asterisk wrote:
Hello Andrew,
Thanks a lot for the detailed response. It's deffinately informative.
I was wondering if you could discuss the IAX -- Ipsec setup you have?
Do you have a box outside of the Asterisk that takes care of the
business
or
On Tue, Mar 23, 2004 at 10:14:36PM -0500, Sean Cheesman wrote:
1. There are VERY clear directions at the bottom of every email on how
to unsunscribe.
2. If you MUST send this to the list, make sure you spell unsubscribe
right.
That could be achieved by copying and pasting the
Hi Craig,
Someone mentioned packet8 to me earlier, having reasonable international
calls from australia, I'd assume they could terminate to it.
On Sat, Mar 20, 2004 at 10:14:44PM +1130, Craig wrote:
Greetings from downunder,
Does anybody know of any organization providing reasonably priced
G'day,
Just as a random side note, IIRC, in Australia a couple of years ago
Telstra lost a law case regarding music on hold.
Google turned up the following url:
www.internetnews.com/bus-news/article.php/38201
This doesn't really answer your question regarding those specific files,
but as part
On Wed, Mar 10, 2004 at 07:37:03AM -0600, Jim Sneeringer wrote:
I have been told that Voicetronics cards are not supported by Asterisk, but
I don't know for sure.
I used to use the VPB4 cards.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Sun, Mar 07, 2004 at 07:04:55PM +0100, Hans-Henrik Andresen wrote:
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
Maybe itneeds SIP/hha1|20,S(10) ?
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Did you create the various /dev/zap devices?
Does * automagicaly uses ztdummy or should I tell him to use it?
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On Tue, Mar 02, 2004 at 09:36:29AM +0100, Micke Andersson wrote:
Hi.
If I want to have all my users (sip) in q mysql
I've tried a few thingies.. but I didn't gett all the needed fields..
like nat, callerid, etc etc
nat can be set globally in the [global] section (funnily enough.)
G'day,
I was wondering if anyone knows if its possible to have conference calls
distributed between multiple * servers? (as opposed to having conferences
on just one server)
I'd imagine this would be helpful in reducing latency with international
calls.
Thanks,
Andrew Griffiths
On Sun, Feb 22, 2004 at 08:21:18AM -0500, John Fraizer wrote:
Use the outgoing call feature of asterisk to have the servers join each
others conferences. It's very simple.
Okay, that makes sense, thanks.
Thanks,
Andrew Griffiths
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Just a silly question, but can the asterisk box ping/contact the client?
(IE, is the routing correct on the * box?)
How can I get asterisk to indicate why it is ignoring SIP REGISTER requests?
P.S. - netstat shows that someone is listening on port 5060
On Sun, Feb 22, 2004 at 09:12:47PM -0800, [EMAIL PROTECTED] wrote:
Just a silly question, but can the asterisk box ping/contact the client?
(IE, is the routing correct on the * box?)
(just to clarify, I mean the natted IP address..)
___
On Sat, Feb 21, 2004 at 09:08:02AM -0500, Bill Michaelson wrote:
Does anyone have any ideas?
Can you check the errno? strerror(errno); should give you a string of why it
failed. (Just be careful not to use other stuff which touches errno after the
fork()
(note, I didn't read over the code,
/wait would be better
I would imagine.
- andrewg
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On Thu, Jan 01, 2004 at 09:48:36PM -0700, Steve Murphy wrote:
Hello--
[snip]
Trouble is, asterisk only sees the brain-dead interface. How do I
exorcise it from the kernel, or at least make the SB the first-priority
one? rmmod didn't seem to do anything. Playing with the Redhat sound
, you might be able to do
like mkdir complete; make PREFIX=${PWD}/complete install; tar czvf install.tgz
install; and copy install.tgz to the machine you're trying to compile on
and extract the install.tgz in /
It helps if you are building on the same distro/patch level as your target.
- andrewg
On Thu, Dec 18, 2003 at 03:02:42PM -0700, Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space (or ventilation) for the monitor and
keyboard. This
As I think was pointed out before, does the phone have a default gateway?
If so, is it correct, if not, that'd be your problem most likely.
Also, does the phone have traceroute?
On Thu, Dec 18, 2003 at 05:09:59PM -0800, Paul Mahler wrote:
In default.cnf for the 7960 I have
# NAT/Firewall
Implementation wise, it would be more frustrating to kick the already
registered user off, and make it more likely it'd be noticed if there
where two registered people.
On Thu, Dec 11, 2003 at 10:15:10PM -0800, Chandra wrote:
last time i was experimenting IAXClient as a true client from dial up
On Sun, Nov 16, 2003 at 08:33:22PM -0800, C M wrote:
hi,
i am getting these errors while installing asterisk. i
reconfigured kernel and i have all the modules
installed.
kernel-source
readline
readline-devel
openssl
openssl-devel
this is the error: (at the last part of the
I haven't looked @ Frrebsd support, but possibly using gmake will fix
the problem pfor you?
On Wed, Nov 12, 2003 at 03:29:58AM -0500, Andrew Joakimsen wrote:
I am trying to get Asterisk to compile on FreeBSD 4.8. Per bug 389, BSD
support should be in CVS. I have also tried applying the patch in
The VPB4 worked for me. The vpb.conf needs to be updated to reflect that vpb4
works.
On Mon, Oct 27, 2003 at 04:34:38PM -0500, Jorge Mendoza wrote:
Hi,
I have two OpenLine4 boards, and would like to test with *.
But I see in vpb.conf that only V6PCI/V12PCI is mentioned. It means that
On Sun, Oct 12, 2003 at 09:01:40AM -0400, Andrew Kohlsmith wrote:
unfortunately you can't do outgoing calls with LineJACK. If you have to
place outgoing calls, then buy some FXO VoIP Gateway (Micronet,
Audiocodes) or digium hardware.
Um, why? is the LineJack not a FXO _and_ FXS
Yes, I have.
Ahh, lets see:
- I grabbed the redhat kernel src rpm, and recompiled.
- I then setup the vpb driver.
- Oh, a big thing. when you get the cards, MAKE SURE the revision is greater
than 19, otherwise it doesn't detect line drops correctly.
- And uhm. lets see. I had problems getting
On Sat, Oct 11, 2003 at 10:25:37PM +0930, [EMAIL PROTECTED] wrote:
Andrew
I am having trouble with
Sound ( only if you dialling from outside )
Hmm. not really. uhm. could play with the volume levels. or it could possibly
be something else.
Cisco phone can not dial out
Sounds more like
On Sat, Oct 11, 2003 at 10:45:26PM +0930, [EMAIL PROTECTED] wrote:
Sound is OK on inside phone
When calling in sounds bad
I don't think I ever got that happening - wasn't part of the scope, so to
speak. How do you mean sounds bad? choppy? it might be converting to/from a
high cost codec.
I was looking into using satelitte for a backup internet connection at one
stage, iirc, its:
- 500ms transmit/recieve latency
- if yours sat connection terminates in the us, you should be able to reach
most place in 30ms
- if you're going to europe (from the termination of the sats in the .us),
I didn't get outgoing calls to work on the linejack cards. Which fortuately,
didn't matter. There where some notes outthere by people who said they got
it working. *shrug* didn't work for me.
If you use GNUGK, iirc, it works fine.
Thanks,
Andrew Griffiths
On Sat, Oct 11, 2003 at 08:24:59PM
On Sat, Oct 11, 2003 at 05:16:11PM -0700, [EMAIL PROTECTED] wrote:
I didn't get outgoing calls to work on the linejack cards. Which fortuately,
didn't matter. There where some notes outthere by people who said they got
it working. *shrug* didn't work for me.
If you use GNUGK, iirc, it works
On Tue, Sep 23, 2003 at 08:54:50AM -0400, costas wrote:
Hi,
I am an experienced developer with Windows and familiar with Linux. I am looking for
a SIP solution.
1) How does Asterisk compare to VOCAL in terms of support.
*shrug*
2) Is Asterisk free?
yeah
3) Where are the docs?
Hi all,
When using trying to dial a number in the US, all I get is a Sprint recorded
voice saying something like Number could not be recoginiised, Please hit 1
then enter the area code you want to dial in, then something or rather like hit
8 for spanish. The card I am using is a Voicetronix
On Wed, Sep 10, 2003 at 05:09:27PM +0200, Zara Trousk wrote:
Hi there,
I?ve been out for some months now, haven?t been checking the list at all.
Does anyone know if the problem with the Quicknet Linejack (FXO) card dial out to
PSTN with asterisk was solved?
Is anybody working on it?
Hi,
Has anyone had a play with the Zultys 4x4 IP phone, if so, did it work fine?
Reference: http://www.zultys.com/summary_ZIP4x4.htm
Thanks,
Andrew Griffiths
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Hi all,
Does anyone have a working IXJ / Dial in config they'd lke to share with me?
Thanks,
Andrew Griffiths
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), its most likely a problem. try switching cards around in
the box.
-andrewg
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On Tue, Sep 02, 2003 at 10:10:17PM -0500, Peter Pauly wrote:
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
correctly from X-lite but nothing else happens - no audio is
heard. My understanding is that I should hear some sort of
I am using x-lite with the asterisk demo
On Tue, Sep 02, 2003 at 09:35:34PM -0600, Gavin Hollinger wrote:
Sounds like an IRQ issue.
Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but
Looks ok?
cat /proc/interrupts
CPU0 CPU1
0: 9228581476395IO-APIC-edge timer
1:
Hi all,
Currently trying to get asterisk to dial out with an Internet Line Jack card,
however, it does not use the pots line, only on the line it dials out of. This
is similar to the previous thread/posting Asterisk won't answer pstn ring,
but I didn't find any follow up to get it working.
My
Depending on your distribution, you will either need to install your own
kernel, or install the kernel source. it should come as a package for your
distro.
On Sat, Aug 30, 2003 at 08:23:08PM +1000, Phillip Britt wrote:
Hi,
I am quite new to Asterisk and Linux in general. When l try to
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
inpin++;
/* XXX Need to do something
Well, considering I replied as soon as I got it, it could always be a delay in
smtp or whatever...
Anyways, the documention advertises a feature which isn't present in the
module it indicates it is in. This would normally be classifed as a bug,
or do you feel it doesn't need to be raised?
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