On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote:
> Does anybody runs Asterisk on AMD64?
>
> I can compile it on Gentoo, and start Asterisk a command line but as
> soon as I connect any device (like Sipura ATA ), asterisk crashes.
Runs well here.
--
Dave Cotton <[EMAIL PROTECTED]>
Tommy,
If you meant VoicePulse, here is how to set it up
http://asteriskathome.sourceforge.net/handbook/index.html#Section_3.3.2
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My motherboard is TYAN Tiger K8W.
I was happy with this board and previous te405p,
except some echo issue.
I will try on 2.6.13 on 955X chipset.
But I am not an expert on linux.
So I want to know easier way.
Any successful installation on FC4?
If someonw know any success story of te406p, please
sha
Hi,
This page
http://linuxcommand.org/man_pages/logrotate8.html
has a sample config file with a file size option
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Watch the Hurricane Katrina Shelter From The Storm concert
http://advision.webev
I don't use Voiceplus so I don't
have the settings. Did you look in the Wiki?
Chris
- Original Message -
From:
Tommy
Denton
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, September 11, 2005 11:50
PM
Subject: [Asterisk-Use
I've been burning the midnight oil trying to configure Asterisk for the
first time. If you have a 2 port SIPURA 2002 can you call from line 1 back
to line two? I have a standard two line, DTMF telephone connected to both
ports. Both lines one and two ARE registered and I can get and leave VM on
Ladies and Gentleman,
I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going.
I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP.
Is there some docs somewhere on how to confg
"Priority" traffic by ARRL standards would fall into both of these
categories. What they are saying is that if someone is in a area where
a ham is operating and needs to get someone hauled out via emergency
services, priority traffic would take precedence over normal traffic.
Not quite a "Mayday"
II noticed that Caller ID presentation is not
making it to my cell phone through outound Telasip calls and I don't know
why. It may very well have been this way for awhile (or always, not sure I
called my cell phone during telasip testing). Does Telasip expect a
different format than SetCI
Try contacting the repeater trustee for http://www.wa3key.com/blura.html.
They have a picture of one on their site with it lit up.
You will need to recrystal the radio to a proper TX/RX
pair for 70cm. However, depending on your area, you should contact your
local repeater coordnator so you
I can understand that. I'm a KL7 call so comms could
mean the matter of someone getting picked up or freezing to
death.
It troubles me that radio site owners (the ones who hold
the pink slip on the tower and hilltop) are not providing power. In AK,
most of these sites are multihomed
with
Hello,
I have been trying to setup a Voxee Sip termination. If anyone has
extensions.conf different than Voxee suggestion.
Can you please send me a copy?
Thanks!
Jerry
Voxee web site advises to use:
[voxee]
exten => _1NXXNXX,1,Dial,SIP/${EXTEN}voxee
exten => _1NXXNXX,2,Hangup
exten
Thanks. I'm afraid that's the case. But I remember
reading somewhere that some syslog variants of syslogd
actually have the file size limit build-in.
--- Brad Hughes <[EMAIL PROTECTED]> wrote:
> Not possible to do natively
>
> Write a script yourself to monior it's size and
> invoke logrotate t
Not possible to do natively
Write a script yourself to monior it's size and invoke logrotate to
rotate it when that size is reached. And cron your script however often
you want (ie 10 times a day?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of YT Lim
Does anyone know how to limit syslog file size?
Logrotate only rotates log files (i.e. irrelevant of
file size), and a log file size can grow extremely
large before it is rotated.
/Y.T.
Do you Yahoo!?
The
Does anybody runs Asterisk on AMD64?
I can compile it on Gentoo, and start Asterisk a command line but as
soon as I connect any device (like Sipura ATA ), asterisk crashes.
--
#Joseph
___
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Asteris
Matt Riddell wrote:
Chee Foong wrote:
i guess may be it's a 64bit variable. so you can only use 0-63.
LOL
Bits work like this
[128][64][32][16][8][4][2][1]
So, you have a whole lot of bits, each one moving from right to left inceases
a power of 2. Say you wanted to represent 1
Chee Foong wrote:
> i guess may be it's a 64bit variable. so you can only use 0-63.
LOL
Bits work like this
[128][64][32][16][8][4][2][1]
So, you have a whole lot of bits, each one moving from right to left inceases
a power of 2. Say you wanted to represent 10, then you would turn on the 8
Well.
That means pci_register_driver probably not ding what it supposed to do.
In newer kernels pci_module_init should be replaced with
pci_register_driver as pci_module_init doesn't it what it supposed to.
How brave are you at getting a new kernel on your system?
I'm currently running on 2.6.13 on
Because the server does not want dead or unconnected phones that might
move.
If the phone does not send a REGISTER every so often, periodically, then
the server will assume the phone is no longer available to send calls
to.
Unlike the Government, Asterisk will not send checks to dead people.
Rac
I modified wct4xxp.c and installed it.
This is the message for 'modprobe wct4xxp'
--
FATAL: Error inserting wct4xxp
(/lib/modules/2.6.9-1.667smp/extra/wct4xxp.ko): No
such device
FATAL: Error running install command for wct4xxp
astpbx kernel: Oops: 0
> I think the biggest thing that hurts ham radio's ability to react to a
> crisis is the lack of equipment and operators. Most of the traffic we pass
> is "Health and Welfare" with "Logistics" being the second to it.
You might be interested to take a listen to the latest ARRL News -
they give a
Olle E. Johansson wrote:
If you are not roaming, set host= and disable
registration in the phone. Then Asterisk will always know where the
phone is.
Not roaming, but it is DHCP based and fixing it would be problematic due
to the way the network is setup.
I'll just continue regularly rebootin
I would like to know if somebody else experienced that:
sip show peers will always drop the first character of the 11th line.
while sip show peers like [0-9,a-z] will not drop any character.
Can anybody test this, please?
bye
Ronald Wiplinger
The latest CVS versions support Presence a lot better.
PaulH
- Original Message -
From: "Trevor Peirce" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, September 12, 2005 8:57 AM
Subject: [Asterisk-Users] Presence Fully Supported?
> I've
You should have just done this:
rmmod wct4xxp
rmmod zaptel
modprobe wct4xxp
It will do the same thing
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jason Kim
> Sent: Monday, 12 September 2005 00:34
> To: asterisk-users@lists.digium
I've seen lots about presence and Polycom phones recently. I've got one
here for evaluation but noticed other phones only seem to appear busy
when they initiate a call. If they receive a call, they still show as
available.
Is this a config problem on my part, or is that as far as presence is
I've got started coding something similar to this. It will have a web
interface and a little for reporting abilities. How soon do you need
this? What I have will be open source when it is done but I've not been
rushing at all.
Darren Wiebe
[EMAIL PROTECTED]
Brian Roy wrote:
On 9/11/05
Is there a situtation where it could hurt to turn off the reset? I have
never had a problem with it cutting off a call or anything but it is
somewhat concerning to see it in the console all the time.
Thanks,
Steve
- Original Message -
From: "JOAO CARLOS MOURA" <[EMAIL PROTECTED]>
To: "A
Just a shot in the dark here.
I bought this unit http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5792377951&rd=1&sspagename=STRK%3AMEWN%3AIT&rd=1 a
couple months ago hoping to connect it to an * system for experimentation.
I am a HAM n00b. I can found no documentation on this unit anyw
Hi all.
Searched the archives but couldn't find anything on this:
I want to track 2nd incoming calls on a single line but don't want to pass the
Call Waiting pips along to the engaged user. IE: I want Asterisk to detect
that CW is currently being transmitted on the line, and track it, but not
Don, I agree with you on many fronts. I come from a radio background
and here in southern cal unless we fall into the sea nothing will take
out all of the communications here including ham because we are not in
low lying flat land and were too diversified, over 150 miles and as
many mountain to
cytrex2*CLI> cdr mysql status
Connected to [EMAIL PROTECTED], port 3306 using table cdr for 3 days, 18
hours, 4 minutes, 3 seconds.
Wrote 62028 records since last restart.
That is what you should see.
-Matthew
> From: alexandre zhang <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List
Hello,
i have succesfully compiled and installed newest channel driver ooh323c with
asterisk CVS-HEAD.
I have small problem - when the asterisk logins to the GnuGK its shchown as
unknown type:
RCF|195.214.XXX.XXX:1720|ASTERIX2:h323_ID|unknown|9681_endp
Sun, 11 Sep 2005 22:34:34 +0200 C(0/0
Time and time again, emergency action drills take place in
cities to target where their weaknesses are in "crisis" handling. Usually
they involve planes crashing or explosions (mock of course). Obviously
they were never prepared for this sort of disaster in their recovery plan.
I've parti
Andreas,
You may like to take a look at
http://mundy.org/blog/index.php
The part of Call Out feature is near half page, the
paragraph heading is "Phone Home" (with ET Image !!!)
I tried it and it work great.
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thanks for ur help
Run the command "cdr mysql status"
I got the following msg
' No such command 'cdr mysql' (type 'help' for help)'
But, I run ' show modules'
cdr_addon_mysql.so is in the list
Do u have an idea about it ?
Thanks
Matthew Boehm <[EMAIL PROTECTED]> 写道：
Run the command "cdr
I will be out of the office starting 12/09/2005 and will not return until
16/09/2005.
Dear Sir / Mdm,
I'm currently on course and are not in office.
During this period of time, I have minimal access to internet and email
cccess. As such, I might not be able to reply to your queries promptly. I
Thank you for all
Sorry my English
Jmoura
- Original Message -
From: "Jason Walker" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Saturday, September 10, 2005 21:40
Subject: RE: [Asterisk-Users] TE110P reset
You are correct. I did not ex
Michael D Schelin wrote:
The two best forms of communications in a real disaster and one always
has been is #1 Ham radio. and #2 satellite telephone. Ham radio is
global and has proven time and time again to be the most reliable when
the infrastructer has been damaged. The U.S government is t
I use the module that this in the [EMAIL PROTECTED] and functions very well.
[]'s
jmoura
- Original Message -
From: Jonathan k. Creasy
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, September 11, 2005 15:00
Subject: RE: [Asterisk-Users] cdr_addon_mysql.so pb
thanks u for ur help
cdr_addon_mysql is built without error. I got the cdr_addon_mysql.so
I input cmd 'show modules' under CLI. I saw cdr_addon_mysql.so .
I built also res_config_mysql.so at same time and it works fine.
Thanks
"Jonathan k. Creasy" <[EMAIL PROTECTED]> 写道：
Did you con
hello asterisk users,
i an using asterisk cvs 1.0.9 in a pIII 733mhz 256MB RAM redhat 9.
i have a TDM400P with 2FXO and 2FXS modules. in my fxs i want to get australian dial tone and for all asterisk operation i want to use australian tones. by default it is US. to change this i have edited followi
Run the command "cdr mysql status" from asterisk CLI. What does that say? If
it says command not found then the module is not loaded.
-Matthew
> From: alexandre zhang <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Mon, 12 Sep 2005 02:11:05 +080
Did you confirm that cdr_addon_mysql is
indeed built and loading? I had missed that it wasn’t being built (I didn’t
have mysql-devel installed) when I first tried to do that a few months ago.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of alexandre zh
> Rich Adamson wrote:
> > Running fc3 with current cvs-head...
> >
> > Is there a nice way to rotate the /var/log/asterisk/messages file without
> > shutting down asterisk?
> >
> > I'm currently rotating the log files via cron, however my script requires
> > asterisk to be shut down, which also k
hi
I load cdr_addon_mysql.so without error
configuration of cdr_mysql.conf
[general]dbhost = localhostdbname = rechargedbuser = rootdbpass = astdbport = 3306dbsock = /var/lib/mysql/mysql.sock
But, I get nothing in the table of cdr of my database.
Somebody have an idea ? Thanks you
Hello,
I have released Ruby Asterisk Gateway Interface (ruby-agi) v0.0.2b.
Any feedback, bug report, suggession, feature request is most welcome.
ruby-agi homepage:
http://www.rubyforge.org/projects/ruby-agi/
Download ruby-agi v0.0.2b here:
http://rubyforge.org/frs/download.php/5965/ruby-agi_v0
On Sun, 2005-09-11 at 08:44, Mike M wrote:
> On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote:
> >
> > Mark Phillips wrote:
> >
> > The suggestion that I have is for various areas to have dedicated civil
> > emergency com units with strategic reserves of fuel (3-4 weeks worth),
> >
On Sun, Sep 11, 2005 at 10:26:38AM -0600, Rich Adamson wrote:
>
> Running fc3 with current cvs-head...
>
> Is there a nice way to rotate the /var/log/asterisk/messages file without
> shutting down asterisk?
>
> I'm currently rotating the log files via cron, however my script requires
> asterisk
Hi,
We have created RAGI (Ruby Asterisk Gateway Interface) for the open
source community so that Ruby and Ruby on Rails can be used to easily
and effeciently create Asterisk-based applications. Examples: IVR,
call center apps, Asterisk management consoles, etc.
RAGI includes a set of objects to
Rich Adamson wrote:
> Running fc3 with current cvs-head...
>
> Is there a nice way to rotate the /var/log/asterisk/messages file without
> shutting down asterisk?
>
> I'm currently rotating the log files via cron, however my script requires
> asterisk to be shut down, which also kills any outstan
On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote:
>
> Mark Phillips wrote:
>
> The suggestion that I have is for various areas to have dedicated civil
> emergency com units with strategic reserves of fuel (3-4 weeks worth),
> battery backups, etc. These units would have links (fib
Running fc3 with current cvs-head...
Is there a nice way to rotate the /var/log/asterisk/messages file without
shutting down asterisk?
I'm currently rotating the log files via cron, however my script requires
asterisk to be shut down, which also kills any outstanding cli sessions
(eg, asterisk -
I've reported this issue as a bug, and learned that a workaround seems to be
disabling callwaiting :(. Since callwaiting was quite helpful on my systems,
I'm not happy with this.
Anyone interested can go to issue tracer and follow if a fix will be
released soon (ID# 0005188).
I really tend t
Matthew Boehm wrote:
I almost had to change my pants when I saw a CVS update this morning
adding T38 frame recognition to asterisk. I kept looking for the code
that complimented this but haven't seen it yet. And there was no bug
reference so I can't help test.
Interested parties can easi
[EMAIL PROTECTED] wrote:
Is there a way to change our dialplan to fail to PSTN in case Dial(*)
reports circuit-busy (but not busy)? I'd like to send to another part of
extensions.conf, where we'd try Dial(Zap). We're already using the n+101
extension to handle the busy condition with the Bus
I modified wct4xxp.c and make clean; make linux26;
make install; reboot;
But the system is not rebooted.
Because the system is in remote office I will check it
next morning.
Could you let me know your linux version, * version
and motherboard?
Thank you Boris.
--- Boris Bakchiev <[EMAIL PROTECTED]
Hello!
Is it legal to use RedirectAction to redirect a call that is waiting in
a queue?
The idea is to have an external application manage a queue via manager
API. The queue
would merely collect calls and play moh.
I've tryed this already but asterisk sends SIP/Forbidden to the channel
in
On Sun, Sep 11, 2005 at 11:29:00AM +0300, Konrads Smelkovs wrote:
> Haven't tried. The install scripts gets May's release and compiles
> with that. I think some serious porting will be nescessary. Anyone?
If you look at the tarball, you can see it has three patches (for
libpri, zaptel and asterisk
Hi, all
Finally got it to work.
TWo problems.
1. Stupid erro on firewall caused authentication failure. BAsically IAX
ports were forwarded to the * box without noting the interface they were
coming on. Thsi is OK when external clients tried to connect, but when I
tried to connect to outside,
On 9/11/05, Andreas Moroder <[EMAIL PROTECTED]> wrote:
Is it possible to use asterisk to call automatically a list of number.
Yes, it's possible. It will require a little effort to do some scripting, but not much. They key to making Asterisk call out, will be using the call files.
Here is th
Use a separate context for each Dring.
dring2 cadence 0,0,0 will identify the primary number not the secondary.
If you want dring1=main number & dring2=distinctive ring num then you
need dring1 as 0,0,0 and dring2 as the alternate cadence.
This context will ignore the calls on the main number if
Hello,
in our public hospital incase of emergency in the night or weekend we
must call many people.
Is it possible to use asterisk to call automatically a list of number.
The numbers should be called in a round-robin way as long as they don't
take up the phone and confirm by digitin in a code.
T
Well.
Try this please (but only if you're running on the latest sources).
Open wct4xxp.c sources and search for pci_module_init
Replace it with pci_register_driver
So the line should read:
res = pci_register_driver(&t4_driver);
That allows you to get the card working on 2.6.13 in almost exactly th
I'm using FC3.
uname -a
-
Linux billitipcc 2.6.9-1.667smp #1 SMP Tue Nov 2
15:09:11 EST 2004 x86_64 x86_64 x86_64 GNU/Linux
zaptel.conf
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
bchan=1-15,
Title: OpenH323-Channel Q.931-Problems with Gatekeeper
Dear Mailinglist-User
currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities.
SIP and ISDN works fine, but H.323 not.
In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeep
zaptel.conf
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
cat /proc/interrupts
---
What kernel are you using?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 7:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TE406p no interrupts
Hi,
I've installed an TE406p, asterisk1.2 on
Martin Allen wrote:
Asking about inserting Asterisk between a 4 line analogue PBX and the
outside world...
The proposed solution (with 4 x FXO and 4 x FXS using 2 TDM400 cards)
will work fine until the asterisk box dies or suffers power failure.
An alternative may be to use 4 Sipura SPA-300
Did it take an interrupt??
Whats does /proc/interrupts say??
Did you check your span= settings in zaptel.conf??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Sunday, September 11, 2005 5:48 AM
To: asterisk-users@lists.digium.com
Subject
Hi,
I've installed an TE406p, asterisk1.2 on tyan opteron
board.
After installation there is no interrupts from TE406p.
Is this board stable?
Should i change * version to 1.0.9?
Regards,
Jason
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Hi.
Am new to this concept but have been requested to add VOIP capability to a
small office phone system.
They currently have 4 standard analog lines running into a PBX feeding 16
phones, with all the usual features,
call transfer
call hold
internal calls
etc.
would the following seem reasonab
Haven't tried. The install scripts gets May's release and compiles
with that. I think some serious porting will be nescessary. Anyone?
On 10/09/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Sep 10, 2005 at 01:25:35PM +0300, Konrads Smelkovs wrote:
> > Isn't billion a HFC PCI card? see lsp
On linux raid:
Linux raid supports hot swapping well. It doesn't care about the
hardware, which is being swapped, much. Obviously, in simple disk
scenario, which is used fot sw raid, only scsi and SATA can be
hot-swapped. Also, make sure that the motherboard supports hot-swap
SATA, i've seen some
Have you read this
article? Its about Sipura 2000 and Asterisk but have much valuable info.
http://voxilla.com/modules.php?op=modload&name=News&file=article&sid=39
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sander
Sent: den 11 september 2005 09
Are you using the Linksys
router as your PPPoE termination or are using the Netopia??
Alex
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk Users
Sent: Sunday, September 11, 2005 3:46 AM
To:
asterisk-users@lists.digium.c
5000-600?
Do you mean 5060? That is the port for 5060. 1-2 is
for RTP.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B.
Asterisk UsersSent: Sunday, September 11, 2005 12:46 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP
Connection Probl
Hello List,
I set up Asterisk for a client. He is using
Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600
and 1-2). For some reson no one from the out side can connect in. I want
to know if anyone had a problem with either Linksys routers or Bell South
bu
you can try to post your sip.conf so someone can help
the sipura spa 2002 works perfectly with asterisk
Sander
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Paul
ConnVerzonden: zaterdag 10 september 2005 23:15Aan:
asterisk-users@lists.digium.comOnderwerp: [Asterisk-Users]
Conf
Try this ip for register something looks wrong with iax.voipbuster.com
I changed it a while ago because i had some dns problems in with my provider
and this ip came up when i pinged now you can't ping to the adress and it's
another ip
register => username:[EMAIL PROTECTED]
-Oorspronkelijk
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