Hi Jonas
Type 'id asterisk' at your command line.
It should return uid gid and all groups the asterisk user belongs to.
Cheers
Bails
jonas kellens wrote:
From voip-info.org :
/Asterisk will look for these files in the /var/lib/asterisk/keys
directory, so copy them there and make sure only
loadzone=us defaultzone=us but it appeasr to make no difference with
this card.
Hope this helps
Bails
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and I never had a complaint again.
They are IMHO physically too light, nothing a chunk of lead wont fix though!
Thanks
Bails
Cheers,
Gordon
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the same kernel?
Here's a sanity check for you: can your system build the module zaptel?
Having said this to other people and received the follow up
~$ m-a command not found
its nicer to say
apt-get/aptitude install module assistant
first.
Bails
aptitude install zaptel-source
m
Device 30c2
Flags: bus master, fast devsel, latency 0, IRQ 220
Memory at d000 (64-bit, non-prefetchable) [size=64K]
Expansion ROM at ignored [disabled]
Capabilities: access denied
Kernel driver in use: tg3
Kernel modules: tg3
Bails
David @ULC
Just to reply to myself and top post in the process.
Its seems this is not the case on CentOS systems my earlier post was the
part of the output from a box running Debian Lenny.
lspci -v on a CentOS-5.1 box shows neither Kernel driver in use or
Kernel modules.
Bails
bails wrote:
Probably
Gordon Henderson wrote:
didforsale.com have just sent me SPAM to the email address I use here.
What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that
I'll never used their services. Morons.
Likewise.
Bails
Gordon
.
Bails
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for my phone config files.
Does netstat -anp |grep :69 show a 0.0.0.0 ?
man page for in.tftpd says it automatically runs for all local networks
on port 69.
Is eth1 not a local network? How do I get tftp to response on eth1?
Thanks,
Jerry
Bails
phone, the hands free mode is the
loudest/clearest/echo free phone i've ever had.
Bails
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From the asterisk CLI do show features
you'll find # is default for Blind transfer
your entry below is commented out, ie has a
;in_front_of_it
hope this helps
Bails
VoIP Newbie wrote:
features.conf has default settings as follows:
;
; Sample Parking configuration
;
[general]
parkext
as 50% of the
upstream rate.
-A.
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Thanks
Bails
make clean
configure
make linux26
make install
perhaps
Bails
Chris Blunt wrote:
Hi Tzafir / List.
Thank you for your reply.
I have run: make clean
Configure
Make
Make install
I get no compile errors, but still the same problems if I try
Unless he's European, if so these messages will stop next Wednesday,
hopefully
Bails
john beaman wrote:
I too was curious about this, so I copied the text into Babel Fish, and this is
the result:
I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my
return. For any
The snom360 DND button forces the phone to give a 480 do not disturb
response.
Bails
--[ UxBoD ]-- wrote:
Hmmm, okay. But surely it will just try and ring the extension? Or do
you mean setup a seperate extension ie.
exten = 1001,1,Dial(sip/1001,20)
exten = 1001,2,VoiceMail([EMAIL PROTECTED],u
Antony Stone wrote:
On Friday 02 March 2007 07:46, Alan Chandler wrote:
On Thursday 01 March 2007 20:33, bails wrote:
plug it in a linux box and tell us what it is please,
generic-usb-audio or what?
Bails
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've
plug it in a linux box and tell us what it is please, generic-usb-audio
or what?
Bails
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've got the base station plugged in, and the handset connected to it,
but it always says pc unavailable.
My system (xp
Disregard lat post, suddenly saw DECT, what is the output of lsusb though?
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've got the base station plugged in, and the handset connected to it,
but it always says pc unavailable.
My system (xp) sees a usb phone
,VoicemailMain([EMAIL PROTECTED])
include = retardonightfix
Bails
J. Oquendo wrote:
So I have my extensions.conf
(http://www.infiltrated.net/exten.stupidity.conf) shortened
in case someone wants to look. Has someone encountered the following?
I've racked my
brain on this for too long...
I have two
Just out of interest are they openwrt compatible?
Bails
John Novack wrote:
I have been using the Linksys BEFRS81 Version 2 8 port router for some
time now, using IAX to and from other Asterisk boxes, and before that
the 4 port version, but discovered that after 18 minutes or so, SIP
traffic
Ronald Wiplinger
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Bails
We use Asterisk Desktop Manager http://adm.hamnett.org/ very
successfully with both debian and windows desktops.
Bails
Ondrej Valousek wrote:
Hi Mitchel,
this looked Very good to me at the first glimpse - then I realized the
client is Windows only :-(
We have Linux desktops here...
Thanks
exten = _XXX,3,Dial(zap/1/${EXTEN},,A(somefile))
bails
Gustavo Berman wrote:
Interesting!
I think this can help for a start (but I don't know how to continue!!):
[incoming]
exten = s,1,Answer()
exten = s,2,Backgroud(enter-ext)
exten = _XXX,1,Playback(enter-name)
exten = _XXX,2,Record(/tmp
http://www.netphonedirectory.com/pap2_dialplan.htm
might help
remember google?
Bails
[EMAIL PROTECTED] wrote:
Thanks Tijl,
That was a nice one but i would like to have my PAP2 programmed with
this dialplan.
This way i can program it for others PAP2's too. I want to have the PAP2
Eric I've had issues with Iaxy's that sound very similar, performing a
full reset following these instructions and re-provisioning them solved
any problems.
http://www.voip-info.org/wiki/view/IAXy
Bails
Erik Anderson wrote:
Greetings - I have recently purchased 2 IAXys. The documentation
rpm -e packagename ?
Rizwan Hisham wrote:
I Installed it using the rpm that is available for download on trixbox
website along with the ISO image. Well actually its on sourceforge.net
http://sourceforge.net. anyways, unlike the ISO image, rpm package
only installs trixbox and asterisk not
I've never used the Aastra but the AT-320's seem to work fairly well, my
only bug with them is there lack of weight, they slide across the desk
to readily.
Bails
James Dyer wrote:
I'm planning to deploy an Asterisk system in our office soon, and am
thinking of using a mixture of Aastra
exten = 888,8,SayDigits(${AMOUNT-DUE})
B
ram wrote:
Hi
thanks for the quick reply
yes as suggested i did the Following modification
exten = 888,1,Read(${CALLERIDNUM})
exten = 888,2,MYSQL(Connect connid 127.0.0.1 http://127.0.0.1 root
password database)
exten = 888,3,MYSQL(Query
The D_key F_key options are AFAIK only available on the 6.X snom firmware.
I have Application-Version:snom360-SIP 6.2.2
the funcionality is great, even the xml minibrowser works correctly for
menus and images.
Overall sound quality seems better than the 5.x images as well
Bails
J
.
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Bails
Doug Lytle wrote:
bails wrote:
Hi all, I'm trying to set some variables absed on a mysql query, but
the value i want to return is .
from everything i read i cant seem to match on it.
eg.
exten = _X,11,GotoIf($[${datavar1} = ]?30:12)
How about:
exten = _X,11,GotoIf($[${datavar1
Bails
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Your cutting the leading dialed number from each box
exten =
_2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _2XXX,2,Congestion
should be
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN},30,r)
exten = _2XXX,2,Congestion
Bails
Tim P wrote:
Here is my
using LTSP nad
running them as a local app, however it uses portaudio with OSS and i
have noticed that different audio modules/soundcards give very different
audio quality.
eg CMIPCI = very good
VIX82XX = very poor
Bails
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Hi Peter, as one of your customers I would ask you not to dissallow g729
on IAX2 as we currently use it extensively.
Bails
Mark Phillips wrote:
Hi Peter,
I don't see any codec allow=blah statements. If your end user has
something like
[gradwell]
disallow=all
allow=gsm
Then you'll
, and Damn
Corporate Liars.
Bails
0
Steven wrote:
First off I am totally annoyed and let down by PC World Business (PCWB part
of the Dixons Group). I ordered one of these babies from them over a month
ago. After constantly chasing them up they finally told me they couldn't
deliver, and have now
Yes its possible, just create different contexts for each organisation.
Bails
Michiel van Baak wrote:
On 13:40, Thu 20 Apr 06, Douglas Garstang wrote:
Does AMP also let you split up each charity so that each only has access to
manage their own content? That seems to me to be a pretty big
backup_asteriskathome.sh imap.agi passwd-maintsetup-samba
bundle-crminstall-AVMB1ISDN passwd-meetme
supercallerid.pl
configinstall-EiconDiva rebuild_zaptel
Bails
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'
== No one is available to answer at this time (1:0/0/0)
Any Ideas?
Thanks
Bails
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Dinesh Nair wrote:
On 03/16/06 04:45 bails said the following:
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
why are all your spans numbered 1 ? surely they should be numbered
1,2,3
Hi all, I've attached a copy of the debug from the Trend, if anyone
cares to look.
I'll probablt get more of a response fromt the list than the automated
response from digium :(
Thanks
Bails
---BeginMessage---
Dinesh Nair wrote:
On 03/16/06 04:45 bails said the following:
Hi whatever
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
zttool always shows
Sync Source: Internally clocked
surely this cannot be correct.
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Anyone know if asterisk supports q931 85 in the uk?
Thanks
Bails
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as these lines are added to extensions_additional.conf
exten = 600,1,Macro(exten-vm,novm,600)
exten = 600,hint,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED])
anyone know why the hint line is there?
does anyone have custom IAX extensions configured thru AMP?
Thanks in advance
Bails
OK I'm answering my own question but if i add a custom extension in AMP
with no dial string.
Then add a dialstring in extensions_custom.conf like
exten = 600,1,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED])
it works
Bails
---BeginMessage---
Hi all, I'm trying to setup a custom
We have 3 existing switches interconnected via dpnss, we need to
integrate asterisk with these switches via a dpnss link.
Any suggestions?
also does anyone have a link to the differences between isdn30 and dpnss.
Thanks in advance
Bails
incoming info.
Anyone have any idea what protocol voipbuster use for incoming calls??
Thanks in advance
Bails
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, plesae dont say post the [EMAIL PROTECTED] forum).
Any Ideas?
thanks in advance
Bails
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. We are
only the customer, but we're expected to do the running around.
Bails
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BJ Weschke wrote:
On 1/26/06, bails [EMAIL PROTECTED] wrote:
Chris Bagnall wrote:
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair
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change IAX2 Protocol Settings DTMF inband to outband
solves it for me.
Bails
/200410.0.0.4 D 255.255.255.255 5060
OK (46 ms
The SNOM360 can make local and outgoing calls fine
all local and incoming calls to the 360 fail and go straight to voicemail.
Executing SetCIDName(SIP/2002-42dc, bails) in new stack
-- Executing AGI(SIP/2002-42dc
is presently in canada with an iaxy
calling thru our * , he doesnt have this issue.
I presume that the ringing tone is generated by the iaxy itself, and
that therefore the one with no ringing tone is faulty.
Can anyone confirm this?
Thanks in advance
Bails
was;
that they could only be sent direct from the far east
We weren't prepared to take the risk, I mean they turned down cash!
Bails
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qualify=yes in both sip.conf and iax.conf, seems to highlight both the
users and trunks who are currently available in FOP
Bails
Pablo Allietti wrote:
On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote:
Pablo Allietti wrote:
hi all i use asdterisk in my company with Flash
an echo of himself. This is only heard on this call, all other calls
are fine.
So where is the echo coming from??
Thanks
Bails
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in advance
Bails
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passwd you need.
Bails
kchase wrote:
the password to login into asterisk
- Original Message -
From: David Uzzell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 3:11 AM
Subject: Re
ulaw): Permission denied
Nov 9 10:37:29 WARNING[2083]: ast_streamfile failed on SIP/2004-6bca
for crim/main-menu.mp3
Output of # file main-menu.mp3
main-menu.mp3: MP3, 192 kBits, 44.1 kHz, Mono
Any Ideas?
Bails
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Bartosz Piec wrote:
bails napisaĆ(a):
Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3
(format ulaw): Permission denied
^
Any Ideas?
Maybe this is a problem with permisions to this file?
It was indeed a permissions problem /var/lib/asterisk
that the FXS module is not getting all the
signalling, as numbers are missing
I have added relaxdtmf=yes to my zapata.conf but this seems not to help
atall.
Could this be a hardware failure?
Thanks in advance
Bails
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Andrew Kohlsmith wrote:
On Monday 07 November 2005 08:03, bails wrote:
I seem to be having some problems with the FXS modules on i, for example
when i dial
90044117XX
Nov 7 13:01:01 VERBOSE[2516]: -- Starting simple switch on 'Zap/1-1'
Nov 7 13:01:05 DEBUG[2516]: DTMF
achieve the
change of context.
Thanks in advance
Bails
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asking because we currently have several sip phones onsite and lots
of remote iax2 users who would like to see availability without dialing.
Bails
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Asterisk-Users
.
I'm asking because we currently have several sip phones onsite and
lots of remote iax2 users who would like to see availability without
dialing.
Bails
Below is a simple perl script that might do the trick. save it to
[something].cgi and most any distribution's apache web server should
with [EMAIL PROTECTED]
if so how? (my head is bruised from repeatedly banging it against the wall)
wishlist
I would love to have this funtionality available from the amportal,
something like add extensions totrunks.
/wishlist
Thanks
Bails
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provide the relevant timing events.
Bails
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Are there any configuration options to allow certain sip/iax accounts
to dial out over specific trunks, and also to stop them dialing out over
other trunks.
Thanks in advance
Bails
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Hey that works great, only problem is how do i configure an outbound
i.e. from asterisk to firefly extension?
Is this possible?
Cheers
Bails
Michael Graves wrote:
I do this all the time. I simply use the Firefly IAX2 soft phone and
don't bother with SIP at all. I forward port 4569 to my
nat whatsoever.
I have read that STUN offers this service, but cannot picture in my mind
how this works, especially with no port forwarding on the nat (I mean,
how the hell do packets traverse the firewall to find the end device).
Any suggestions welcome
Cheers Bails
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