On Thursday 23 Feb 2006 20:34, Colin Anderson wrote:
It's stupid. Don't ever connect 2 different building with copper.
Just wait until you get some kind of lightening hit or electrical
fault, but make sure you are no where near it. Use fibre.
Thanks for the reply. Unfortunately, the conduit
On Thursday 23 Feb 2006 17:30, Colin Anderson wrote:
I have to provision several dozen * users to a seperate building on our
campus in the same subnet. Ordinarily, I'd just run a gigabit cat6 cable to
another switch if it doesn't violate the 100 metre rule, but this building
is several hundred
On Tuesday 21 Feb 2006 23:16, Chris Bagnall wrote:
£40! That would be a cheap and nasty switch with no prospect
of any management. A managed switch is worth its weight in
gold, /especially/ when you have to look after things remotely.
How does one justify the extra cost of a managed switch
On Tuesday 21 Feb 2006 19:55, Chris Bagnall wrote:
I agree with most of Raymond's other points, but I have to take issue with
this one:
1) If it doesn't support PoE I won't implement it. Support
phones with wall-warts or bricks is just a added hassle and
adds TCO as most end up being
On Thursday 16 Feb 2006 22:20, Jean-Louis curty wrote:
hi,
My question is may be a bit out of scope but I don't know where to turn,
I have a 1760 with a ccme 24 user licences 1 bri card.
I want to configure a bri card in a cisco 1760 on port 0/0,
the card is new, seen by the router, show
On Monday 13 Feb 2006 21:21, Chris Bagnall wrote:
Hello all,
I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of
them: codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes
long from
On Sunday 12 Feb 2006 01:04, Kevin P. Fleming wrote:
[...]
(and by 'you' I don't mean you specifically, Michael, I am referring to
those in this thread who think we should change the code used to write
to every file we write to in Asterisk)
I assume you are referring to me. I did not say that.
On Sunday 12 Feb 2006 10:57, Bob Goddard wrote:
On Sunday 12 Feb 2006 01:04, Kevin P. Fleming wrote:
[...]
(and by 'you' I don't mean you specifically, Michael, I am referring to
those in this thread who think we should change the code used to write
to every file we write to in Asterisk
On Saturday 11 Feb 2006 00:10, Kevin P. Fleming wrote:
Warren Burstein wrote:
How about if it would set a global variable before each disk write so
the SIGFSZ handler would know which file caused it?
Ha!
Signals are asynchronous. This global variable would to be
lock-protected, would
On Friday 10 Feb 2006 15:02, Steve Totaro wrote:
Hello,
For some reason I cannot get Cepstral to work with 1.2.4. I followed
all the directions here
http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
When I try to load asterisk I get
[app_cepstral.so]Feb 10 04:58:36
On Thursday 09 Feb 2006 16:01, Kevin P. Fleming wrote:
Dov Bigio wrote:
Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Maybe a more user-friendly message could be logged, infoming which file
reached the 2.0GB.
On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote:
Hi All
I am having problems with Directed Call Pickup in Asterisk 1.2.1
If extension 100 is ringing, a user at another extension is supposed to
be able to dial *8100 and pickup the call to 100. It isn't working for
me and I cannot
On Thursday 26 Jan 2006 16:50, Michaël Gaudette wrote:
Hi,
I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've
noticed that the CDR logging in MySQL (on a different computer) has
stopped. I thought it wasn't logging anything at all, but I realized after
a bit of searching
On Thursday 26 Jan 2006 20:00, Michaël Gaudette wrote:
Yes I did. Fair question. I think it`s working, but is there anyway to
know for sure? Show modules show app_cdr.so as existing...
Please do not top post.
Please do not post as HTML.
You should be looking for cdr_addon_mysql.so.
The
On Tuesday 24 Jan 2006 22:51, Kevin P. Fleming wrote:
Terry Gilsenan wrote:
Can we get that IP blocked in the postfix access list at digium?
(lists.digium.com [69.16.138.164])? Ozemail are not what you would call
active in stopping spam.
The list had no incoming spam filtering for some
On Monday 09 Jan 2006 13:46, Sven Fischer (support) wrote:
On Saturday 07 January 2006 02:30, Philipp von Klitzing wrote:
Hi!
Now, one user, not the receptionist, has gone in and set his personal
numbers to these function keys thinking that DESTINATION meant setting
a number to dial
On Friday 06 Jan 2006 00:46, A_ Navone wrote:
make[2]: *** [obj_linux_x86_r/simph323] Error 1
make[2]: Leaving directory `/usr/src/openh323/samples/simple'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/usr/src/openh323'
make: *** [optshared] Error 2
any idea ?
None unless
On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
Mike Fedyk wrote:
Matt Riddell wrote:
I would instead recommend the SuperMicro 1U servers - we have had a
really
great run with these.
Do you use Opteron or Intel?
I would not suggest that Supermicro are in Intel's pocket, so they
On Friday 06 Jan 2006 15:44, Walt Reed wrote:
On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:
On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
Supermicro do not do Opteron (or Athlon64) systems.
Supermicro DO do Opteron.
Model numbers please? Searching through
On Friday 23 Dec 2005 08:03, Tomislav Parcina wrote:
I have Grandstream Budge Tone 102 with Software Version:Program--
1.0.5.18Bootloader-- 1.0.0.21HTML-- 1.0.0.42VOC-- 1.0.0.7.
I'm planning to upgrade it with Firmware 1.0.6.7.
My question is, does anybody has any ishues with
On Friday 23 Dec 2005 12:59, Rich Adamson wrote:
This is another thing: Linux tends to use the availble free memory for
IO buffers, disk cache and such. So in the output of 'free', look at the
second line.
I'm not the OP, but for those of us that are not considered strong sys
admin's (but
On Friday 23 Dec 2005 14:38, Tomislav Parcina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
There is a major registration bug with all of the BT releases
which Grandstream are refusing to fix.
Can you please tell me more about that problem?
From a previous reply...
On Monday 28 Nov 2005 15:39, Frank McCarthy wrote:
Does anybody know where I can download ringtones for Cisco 7960's? Need
to be .pcm files.
Download the ringtone generator from Grandstream and make your own.
B
___
--Bandwidth and Colocation
On Monday 28 Nov 2005 16:29, Tony Hoyle wrote:
Noc Phibee wrote:
Thanks sergio for your answer.
But cisco france say me that i cant' bye SmartNet contract on this
product.
You can, but only in the US I believe. I've never found any deal less
than £150 (UK).
I guess that should depend
On Monday 28 Nov 2005 20:41, Tony Hoyle wrote:
Bob Goddard wrote:
It's not so bad... you do get access to firmware to all cisco devices
with that, so if you have more than one device it becomes worth it.
And it is also illegal.
Not true - that's the *point* of the more expensive contracts
On Monday 28 Nov 2005 20:42, Tony Hoyle wrote:
Bob Goddard wrote:
I guess that should depend as to whether it is hardware or software only.
AFAIK all smartnet are software only... I've never heard of a hardware
contract.
No, the vast majority of the smartnet contracts are hardware
On Tuesday 01 Nov 2005 22:25, Robert Rozman wrote:
Hi,
I have Grandstream 100 as only ever present extension for my Asterisk AMP
home setup. Incoming call comes to ring group and then proceeds to
voicemail.
But Grandstream 100 occasionally loses registration (have anyone found any
On Friday 21 Oct 2005 15:26, Jayson Smith wrote:
Hello,
Ok, so where does Goiax related traffic belong? Should Goiax have its own
mailing list? I would tend to think so, but lacking such a specific list,
this list is probably a place where many users and potential users will be
subscribed.
On Monday 17 Oct 2005 04:11, Kevin P. Fleming wrote:
Samy Antoun wrote:
[context1]
exten = s,1,Answer
exten = s,2,SetVar(MYVAR=1)
exten = s,3,Goto(context2,s,1)
[context2]
exten = s,1,NoOp(${MYVAR})
The NoOp in context2 will return 1?
Variables are set on the channel itself,
On Tuesday 11 Oct 2005 23:57, Lee Howard wrote:
Bob Goddard wrote:
On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
Tom Rymes wrote:
Use the right tool for the job!!!
Use a hardware based DSP for faxing not software based.
Why is a soft-DSP to be considered any less-capable than hardware
On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bob Goddard
Sent: Tuesday, October 11, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
On Wednesday 12 Oct 2005 14:54, Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Tuesday, October 11, 2005 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Tuesday 11 Oct 2005 12:19, trixter http://www.0xdecafbad.com wrote:
[...]
It started with the UK as an *example* and everyone seems to have
latched onto that. I wanted to know more than the UK, I wanted every
country. astbill seems to have that data, I seem to have located all
the little
On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
Tom Rymes wrote:
Frankly, I would recommend that you forget about trying to fax with
Asterisk. Buy a good Multitech analog modem and install HylaFAX.
Use the right tool for the job!!!
Actually, you can use HylaFAX and Asterisk together.
On Tuesday 04 Oct 2005 05:17, [EMAIL PROTECTED] wrote:
On Mon, 3 Oct 2005, Aryanto Rachmad wrote:
I sent an email to Digium support and got only a reply like this:
Although the card is being shown as an 'Unknown Device', it should still
work properly.
To be honest, I am not happy with
On Monday 03 Oct 2005 08:51, Olle E. Johansson wrote:
Paul Conn wrote:
I’m receiving the following error over and over, adnauseam:
Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce
received from ‘CNAME-CID sip:[EMAIL PROTECTED]’
Does anyone know what “stale
On Saturday 17 Sep 2005 23:20, Kevin P. Fleming wrote:
Damon Estep wrote:
I do not have the r option in the MOH class, but the files are played
in an order I can figure out, they do not appear to be random either,
same pattern repeats.
Oh come on, its obvious :-)
Have you figured it out
On Sunday 18 Sep 2005 15:15, Francois Meehan wrote:
Hi all,
I have bought an Aastra 480i phone.
In order to configure the phone for using a TFTP server, I had to enter
the TFTP ip address directly in the phone, and then reboot the phone
again.
Is it possible to configure a DHCP server so
On Monday 12 Sep 2005 21:53, Colin Anderson wrote:
The one I like:
http://www.rhetorical.com/cgi-bin/demo.cgi
is toast. I think they went broke or got aquired by someone. Also, is there
a Festival voice that sounds as good as Rhetorical or the AT T stuff? The
According the UK Companies
On Friday 09 Sep 2005 18:10, Tony Hoyle wrote:
Olle E. Johansson wrote:
SIP phones need to re-register every once in a while to tell the server
where it can be reached. If you have a soft phone on a laptop that you
move from network to network - home, office, airport, Barnes Noble etc
-
On Friday 26 Aug 2005 14:54, Julian Lyndon-Smith wrote:
I cannot reach voip-info - is it just me or is the site not available ?
There is a bad route being propogated.
B
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
On Wednesday 24 Aug 2005 09:44, razza wrote:
I have a standard BT home DSL, which means I cannot have a static IP
address, therefore i'm forced to use NAT, I subscribe to a DDNS service
and have written a VB app which polls the router every 10 seconds and
updates the DDNS if appropriate.
On Wednesday 24 Aug 2005 13:40, Kamran Ahmad wrote:
hello
i m getting follwing messages in asterisk-1.0.9 after
small interval. And i have to restart asterisk because
after these errors asterisk cannot do any call. what
is the reason calls are not going out. can u pls tel
me how to solve
On Wednesday 24 Aug 2005 17:22, Goran Dj. wrote:
I'm trying to compile chan_capi-0.5.4 on Slackware 10, but I have bunch
of errors.
(By the way, can I use chan_capi for ISDN card with winbond w6692cf
chipset?)
I'm not a linux expert, still :-)
Before compiling, when I type modprobe capi to
On Monday 22 Aug 2005 04:30, root linux wrote:
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
You do not appear to be in the US but Malaysia. Not sure what these
should be.
My zapata.conf config: -
On Monday 15 Aug 2005 15:19, Tom Tobias wrote:
I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the
stable asterisk build. Both packages configure and compile with no
problems. However when compiling chan_h323 from the
asterisksource/channels/h323 directory I get this
On Saturday 13 Aug 2005 07:29, Eric Bishop wrote:
Hi all,
Anyone able to remotely reboot their password protected Sipura
SPA-3000 from command line. I am trying:
Sipura SPA-3000 from command line:
# wget http://admin:[EMAIL PROTECTED]/admin/reboot
The strange thing is it works fine when
On Tuesday 09 Aug 2005 17:26, Jonas Arndt wrote:
Hi Guys,
I hope this is the correct mailing list for this question.
I have a dual 1.6 Ghz Itanium with 4 Gb of memory. Yes, a lot of power
for Asterisk. I am running SuSE Enterprise Server with the
2.6.5-7.97-default kernel. I have just
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Goddard
Sent: 06 August 2005 23:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setup faxing with latest CVS
On Saturday 06 Aug 2005 22:41, Erick Johnson wrote:
I have been trying to setup faxing
On Saturday 06 Aug 2005 22:41, Erick Johnson wrote:
I have been trying to setup faxing with a recent CVS-HEAD. I have
downloaded and compiled spandsp-0.0.2pre18 and gotten apps_makefile.patch,
app_txfax.c and app_rxfax.c
I'm not suprised that the patch failed. Does anyone know what changes
Please stop asking the same questions over and over.
On Monday 25 Jul 2005 02:46, Balgansuren.B wrote:
Hello,
I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.
1.I couldn't find Asterisk version using asterisk -V command.
How can I
On Friday 05 Aug 2005 17:08, Neil Cherry wrote:
I'm seeing all sorts of problems and it's probably more of my lack
of experience than anything else. I have a BT100 running 1.0.6.7
code. When I go to the status page it says it's not registered
(hmm, that's not good). I also can't get dial tone
On Friday 05 Aug 2005 19:16, Michael George wrote:
I have a pair of snom 360s at a customer and they were giving me Low Memory
errors. The distributor suggested updating the firmware. I did that, to
the one just below 4.0 (which wasn't released yet). One of the phones is
still giving the
On Wednesday 03 Aug 2005 18:46, Michael D Schelin wrote:
Why do you put me down? I have not done a thing to you and I'm not a
spammer. Please stop this activity It's not professional. If I were to
give you bad service please feel free to comment negatively but I've
never dealt with you nor do
On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote:
before I accuse somebody to overbill I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers:0.7269 or 0.2929 ???
Assuming it is per minute;
930 *
posted what.
930 * 0.02 / 60 = $0.31
Whereas, in our industry, we need to compute rates with far more
decimal places than just 2.
- Waldo
On Jul 28, 2005, at 10:31 AM, Adam Dobrin wrote:
Bob Goddard wrote:
[...]
930 * 0.0189 / 60 = 0.29295
I get .31$. Where did you all go
On Wednesday 20 Jul 2005 18:31, chris wrote:
hi kevin,
i tried removing the enitre asterisk directory and upadatesd my cvs folder.
and try to run make.. i'm getting
make_version_h : cannot execute error
The file requires that /bin/sh exists, aside from checking that, try and
execute the
On Wednesday 20 Jul 2005 20:34, Matt Loretitsch wrote:
I wish someone would just post a sample extensions.conf so I could
FINALLY understand this. Could you post at least the hint portion of
yours? I have tried this repeatedly without success and am starting to
feel like a true idiot.
Is
On Tuesday 19 Jul 2005 14:45, Martin Sutherland wrote:
Why didn't I think of using that command...
It shows all - for G729a which is presumably why I'm having a problem
I have purchased 20 licenses from Digium, downloaded binary, registered the
binary correctly, placed it in the correct
On Thursday 14 Jul 2005 17:32, Rob Danz wrote:
Yes, the permissions are okay for getting to that folder.
/var/spool/asterisk is writable (voicemail works that's a subdirectory
under the same path that has the same permissions as the subdirectory
'asterisk-fax'
As the same user that runs
On Wednesday 13 Jul 2005 16:19, Kib Eki wrote:
Hi,
i am running * 1.0.9 with a newer version of the TE405P.
Modprobe wct4xxp and ztcfg are OK.
zap show channels only shows me the following.
my zapata.conf:
[...]
Why can't i see or use my channels?
You're not going to get anywhere unless
On Tuesday 12 Jul 2005 15:51, Colin Anderson wrote:
Thanks for replying. Frustrating, didn't work. Set it to update
automatically, and made an HTML page consisting of:
html
pre
bootloader:
firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin
/pre
/html
Added the HTML page to a
On Tuesday 12 Jul 2005 19:02, Patrick Friedel wrote:
Bob Goddard wrote:
There are 2 problems here, the first is if you click on memory and
the connection count is not 0, then you will be unable to reboot the
phone, all you can do then is power cycle it.
Secondly, to update the phone, you
On Monday 11 Jul 2005 05:02, Michael Stearne wrote:
On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote:
Thanks William and John, I'll look again for that download. Comments
below...
--On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett
[EMAIL PROTECTED] wrote:
FWIW? I bought that
On Wednesday 06 Jul 2005 06:07, wei li wrote:
Hi there:
I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4
box. When I tend to install the addon for mysql CDR billing, It always
return me the following errors:
SIP# gmake clean
rm -f *.so *.o .depend
gmake -C format_mp3
On Friday 01 Jul 2005 16:43, Zoltan Szecsei wrote:
Bob Goddard wrote:
On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first
On Friday 01 Jul 2005 13:08, Terry Wade wrote:
Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
Nope, I doubt that. The end user should read /usr/src/linux/README.suse
and see how to prepare the kernel for building thirparty modules.
-Original Message-
From: [EMAIL PROTECTED]
of /boot at least.
Bob Goddard wrote:
On Friday 01 Jul 2005 13:08, Terry Wade wrote:
Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
Nope, I doubt that. The end user should read /usr/src/linux/README.suse
and see how to prepare the kernel for building thirparty modules.
-Original
of the generated tiff file was wrong.
Interestingly though, when I try to fax out the PRI to one of our
own DDI's, that to say it come back in on the PRI, the fax software
just sits there looking stupid!
On 6/30/05, Bob Goddard [EMAIL PROTECTED] wrote:
I've a stock RH9 system with spandsp 0.18. Faxing out
I've a stock RH9 system with spandsp 0.18. Faxing out over a PRI to a
USRobotics modem on a stock Suse9.3 system with hylafax fails with the
following errors in the hylafax logs:
Jun 30 19:28:53.23: [ 608]: RECV/CQ: Bad 1D pixel count, row 0, got 595,
expected 1728
Jun 30 19:28:53.23: [ 608]:
On Thursday 23 Jun 2005 17:39, jltaylor wrote:
I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?
Yes. 4BRI cards in 2 separate systems hosting 10 nodes.
B
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Friday 17 Jun 2005 18:05, Manuel Casal wrote:
Marco Parmeggiani escribi:
Manuel Casal ha scritto:
I made the make menuconfig and make dep in the kernel sources.
i do not remember well how i solved that problem but i'm sure that
make dep will issue you a warning and stop.
run make
On Friday 17 Jun 2005 17:15, Yousef Herzallah wrote:
Hi,
This error I got it just when I gonfigure zaptel support isdneuro 31
channels.
But if I configure zaptel to support T1 and just 24 channels I have no
problem.
On Thursday 16 Jun 2005 09:25, Mark Brown wrote:
Hi Everyone,
I'm using Asterisk, actually [EMAIL PROTECTED] 1.1 with all Grandstream 102
phones.
NAT is not an issue as all including the server have public IP's
The problem is that the phones keep losing registration with the server.
I
On Tuesday 14 Jun 2005 09:16, David Masure wrote:
Hi,
I'm facing something strange but maybe I haven't the right solution.
What I want ot do is :
Someone from outside call my phone number, I check some informations
using an IVR script and then I want to transfer the call to an external
On Tuesday 14 Jun 2005 11:45, Bob Goddard wrote:
On Tuesday 14 Jun 2005 09:16, David Masure wrote:
Hi,
I'm facing something strange but maybe I haven't the right solution.
What I want ot do is :
Someone from outside call my phone number, I check some informations
using an IVR
On Tuesday 14 Jun 2005 14:30, Bryce Chidester wrote:
I used to use the following but Festival is such a load hog I just
NoOp the same info and read off the console.
exten = 789,1,Festival('You are currently calling into context: $
{CONTEXT} as name: ${CALLERIDNAME}. number: ${CALLERIDNUM}.
On Sunday 12 Jun 2005 08:56, trixter http://www.0xdecafbad.com wrote:
On Sat, 2005-06-11 at 13:47 -0700, Daryll Strauss wrote:
On Sat, 2005-06-11 at 13:10 -0700, trixter http://www.0xdecafbad.com
wrote:
Look at 'big evil corporations' like apple. They did in a year with
mach what the
On Sunday 12 Jun 2005 16:10, trixter http://www.0xdecafbad.com wrote:
On Sun, 2005-06-12 at 15:06 +0100, Bob Goddard wrote:
On Sunday 12 Jun 2005 08:56, trixter http://www.0xdecafbad.com wrote:
On Sat, 2005-06-11 at 13:47 -0700, Daryll Strauss wrote:
On Sat, 2005-06-11 at 13:10 -0700
On Friday 10 Jun 2005 22:46, list wrote:
RFC 1912
Every Internet-reachable host should have a name. and then For every IP
address, there should be a matching PTR record in the in-addr.arpa
domain. and Failure to have matching PTR and A records can cause loss
of Internet services similar to
On Saturday 11 Jun 2005 14:56, Tracy Phillips wrote:
[...]
I wonder if there is an RFC from top posting? I doubt it... seems the
rest of the world can get along fine reading top posts...
rfc1855 details the netiquette guidelines.
From paragraph 3.1.1
If you are sending a reply to a message
On Thursday 09 Jun 2005 23:45, Andrew Kohlsmith wrote:
On Thursday 09 June 2005 13:15, Bob Goddard wrote:
The Via processors emulate the i686 just fine. The problem has always
been with GCC.
Got some proof of that? It's generally regarded as common knowlege in
these circles that the via
On Thursday 09 Jun 2005 17:29, abel wrote:
Seshu,
Are you working on a VIA based motherboard?
I am working on a VIA based motherboard.
Andy Powell (author of Asterisk Live! distro) tells me that VIA is not
quite good when emulating i686 behavoir and since his distro is compiled
for i686...
On Thursday 09 Jun 2005 18:55, Kanuri, Seshu (Company IT) wrote:
Abel,
I am working on Intel boards only.
I have tried VIA boards and I do not recommend anyone to work on VIA
boards for a production system. The reasons for this being that there
are just way too many issues with these
On Monday 25 April 2005 08:56, Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk. Does
anybody know how does one go about modifying the C code of Asterisk? Please
do reply.
How many times must you ask this?
If you do not know how to
On Wednesday 25 May 2005 11:27, Ronald Wiplinger wrote:
Bob Goddard wrote:
On Monday 25 April 2005 08:56, Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk. Does
anybody know how does one go about modifying the C code of Asterisk
On Wednesday 25 May 2005 10:15, Asterisk User wrote:
Hi all,
I have problem with my Asterisk.
I'm using the softphone Xten-Lite. I've removed the SIP client information
in sip.conf. The softphone can't register to Asterisk, but it can make
outgoing calls.
I've tried to add back the SIP
On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote:
Hello,
I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
a register staement (nothing in thertereal log). With the 1.0.3.81
version, the phone
On Tuesday 24 May 2005 18:16, hank smith wrote:
they ever going to fix it?
I sure as hell hope so. Such a bug is a show stopper.
B
- Original Message -
From: Bob Goddard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Tuesday 24 May 2005 17:07, Daniel ANDRE wrote:
Bob Goddard a écrit :
On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote:
Hello,
I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
a register
On Thursday 28 April 2005 19:49, Kristian Kielhofner wrote:
Hello everyone,
I don't know if it is just me, but I can never get a connection out of
Digium's FTP server. I can connect and login just fine, but both active
and passive ftp timeout before I can even get a directory
On Saturday 23 April 2005 19:13, Matt Klein wrote:
$4,172.38 USD and I'll programin anything you want for asterisk server.
You are too stupid for the job.
On Sat, 23 Apr 2005, Franz wrote:
PLEASE CAN SOMBODY HELP ME PROGRAMIN AN VoIP ASTERISK SERVER
Atentamente,
Franz Schuverer Arrue
Has anyone managed to find what the maximum concurrent faxes, either
incoming or outgoing using SpanDSP? I realise that it depends on what
system it is running on, so some basic details of the system would
be appreciated.
B
___
Asterisk-Users mailing
On Wednesday 13 April 2005 19:39, jeff oconnell wrote:
i'm running gentoo on a via motherboard ( MII6000, Samuel 2 ):
uname -a
Linux 2.6.11-gentoo-r4 #5 Fri Apr 1 19:26:50 EST 2005 i686 VIA Samuel 2
CentaurHauls GNU/Linux
and asterisk from cvs-head is core dumping:
On Monday 11 April 2005 15:15, Jesus Mogollon wrote:
Hi all
I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to
configure both NICs with bonding enable (bonding miimon=100 mode=1). I know
certain features (like load balancing) under a bonded configuration is not
On Monday 11 April 2005 22:36, Tim Connolly wrote:
I'm assuming I would see an error if this was bad:
ldd /usr/lib/asterisk/modules/chan_zap.so
linux-gate.so.1 = (0xe000)
libpri.so.1 = /usr/lib/libpri.so.1 (0xb7f89000)
libtonezone.so.1.0 =
On Friday 01 April 2005 23:03, Tim Bass wrote:
Congratulations!
Tom Ivar Helbekkmo and Francesco Peeters
Voted Number One Bullies of Asterisk-Users.
You are the bully. So far the majority wish the email list to
continue and yet you still continue to demand that Digium
convert to a forum.
On Thursday 31 March 2005 23:11, Tim Bass wrote:
The UNIX Forums have over 28 thousand registered users. I have many
years of experience in both email lists and on line forums and I can tell
you without a doubt that on-line forums are far superior to email lists.
There is no comparison.
On Friday 01 April 2005 04:28, Joseph Gutowski wrote:
Ok, since I guess no one else wanted to bite -- I will.
I installed PingPlotter, switched to UDP just to be the same as you,
and ran it against sip.broadvoice.com. Absolutley no problems, no
packet loss at all.
Ran it with all of the
On Tuesday 29 March 2005 14:08, Rikard Westlund wrote:
[...]
When I start Asterisk(asterisk -vc) I get this:
Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No
D-channels available! Using Primary on channel anyway 16! == Primary
D-Channel on span 1 down
[...]
I'll
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