Thank you very much. That has done it :-)
Jeffrey C. Ollie wrote:
On Tue, 2005-04-12 at 20:29 -0700, Steven P. Donegan wrote:
I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux
(2.6.10 kernel patched as suggested). I get compile warnings and
modprobe failure on zaptel stuff
I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux
(2.6.10 kernel patched as suggested). I get compile warnings and
modprobe failure on zaptel stuff:
zaptel: Unknown symbol crc_ccitt_table
I'm assuming that something needs to be in the kernel space that isn't -
any pointers to
Is there any support in Asterisk for encryption of IAX and/or any other
VOIP protocols? I haven't seen anything on this in the wiki or on the
list. Just curious.
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PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven P.
Donegan
Sent: Friday, February 04, 2005 8:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Encrypted VOIP?
Is there any support in Asterisk for encryption of IAX and/or any other
VOIP protocols? I haven't seen anything
Well, not really 'looking for' anything except info - as an earlier
gentleman said I can always just do stuff over a VPN - it's more a
simple interest in where things might be going.
Thanks as always for the feedback folks
Nick Bachmann wrote:
[EMAIL PROTECTED] wrote:
If I remeber
Hmmm :-)
OK - grab a current RedHat Fedora Core 3 set of ISO's, install - about 2
hours... Do the up2date gig - time depends on your network connection.
Do a cvs based asterisk install - about 30 minutes (more than happy to
share the scripts to do this). Total elapsed time 24 hours max. This
Is there any way to configure a zap channel to wait for some period of
time or number of rings before answering the line? I would like to have
a line shared between in-house emergency phones and the asterisk PBX.
Thanks.
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Today I had channel 1 on my TDM400 go to sleep, verified by pluggin in
known good telco lines in various combinations on channel 1 through 4 -
problem is channel 1, not anything external. So after seeing lots of
stuff on the list re: TDM400's I power cycled, removed board and let
linux say
I don't want to be negative here, but I do believe people who go to do
this know the potential risks they face. In many countries (4 of which I
have direct, although several year old experience with - all in Asia)
taking a local phone line and attaching asterisk to it and gatewaying
traffic
Funny, the only thing I addressed was the direct threat of busting the
contract/acceptable use policy of your Telco/local government. I didn't
go anywhere near the other risks:
1) you mess up your extensions.conf and some bozo - on purpose or
otherwise - runs up some insane bill on that nice
OK, I have done all the stuff at my end and at Bellsters end to add 21
new area codes (all of california) to the Bellster dial plan. Pretty
cool deal! I hope others go for this quickly - as it could be a really
nice co-op.
I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk
Agreed - the announcement is not needed - although it's kind of neat -
perhaps it could be something that was optional/configurable from the
bellster web page?
Nathan Goodwin wrote:
I love belster, I added a route for the 518 area code, (that covers
most of upstate NY), only thing I wish I
Be very careful with your 4801 - Soekris boards are designed to only
support 3.3V PCI at very low power levels - putting a TDM card in there
would very much exceed the allowed power use on the PCI connector.
My setup will be using my Soekris 4801, a 40G 2.5 IDE drive for
voicemail storage/boot
On the same general subject, Asterisk users get-togethers, who might be
interested in sharing conversation in the Disneyland/Knotts Berry Farm
area in Orange County, California?
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richard wrote:
I have the following line in my extensions.conf which when I dial 100
(from my BudgeTone 100, and then wait a few seconds) I will get an
outside line.
exten =100,1,Dial(Zap/1,20)
What do I need to put in the extensions.conf file so that I can dial 9
(and then a number) and then
Michael Graves wrote:
Hi All,
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my * server
to be more like an appliance than a PC. However, the
Rich Adamson wrote:
Current experience... three spa-3000's are far more stable then a TDM
card, and you'll get three fxo's plus three fxs's for less money.
Except for the little problem I've fought for about a week without any
Joy - no combination of efforts from numerous sources (wiki, this
Nabeel Jafferali wrote:
Except for the little problem I've fought for about a week
without any Joy - no combination of efforts from numerous
sources (wiki, this forum members, my efforts) has succeeded
in a spa-3000/asterisk combination that actually works. If
you have specific spa-3000 and
Nabeel Jafferali wrote:
Dec 31 12:03:55 NOTICE[7145]: chan_sip.c:7486 handle_request:
Failed to
authenticate user WIRELESS CALLER
sip:[EMAIL PROTECTED];tag=83eaec7dcb80f5feo1
Have you tried the A prefix trick, which uses Line 1 Call Forwarding
as opposed to PSTN Line Call Forwarding (with the
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to
authenticate
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system - however they don't get routed properly. The Asterisk message:
Dec 30 07:47:16 NOTICE[2745
Michael Graves wrote:
On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when incoming
calls arrive on the telco port they arrive properly on the Asterisk
system
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
I have a Sipura 3000, apparently configured correctly, when
incoming calls arrive on the telco port they arrive properly on the
Asterisk system - however they don't get routed properly
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
The Sipura has registration entries in sip.conf for both lines - and
from my earlier post appears to register just fine. I'm still
clueless on the failure originally reported.
Steven,
So, of the 1001, 1002, 1003, etc. one of those
The below is a asterisk message when I try to call from a callerid
blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not
consciously put any restrictions on incoming calls...
Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to
authenticate user WIRELESS CALLER
Kristian Kielhofner wrote:
Steven P. Donegan wrote:
The below is a asterisk message when I try to call from a callerid
blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not
consciously put any restrictions on incoming calls...
Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486
I recently swapped 2 FXO modules on to what had previously been a 4 FXS
version of the TDM400 board. The FXS ports are recognized - the FXO
ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both
say channel 1 isn't there). Has anyone experienced this problem? All
software is
they are placed on your TDM400.
Best regards
Lamine
- Original Message -
From: Steven P. Donegan [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, December 27, 2004 6:01 PM
Subject: [Asterisk-Users] TDM400 problem
I recently swapped 2 FXO modules on to what had previously been a 4
Teodor Georgiev wrote:
I have Asterisk (the yesterday CVS) installed on FC3.
No issues so far.
On Thursday 16 December 2004 12:48, [EMAIL PROTECTED] wrote:
Hello,
Since FC3 has been a very recent release
I was just wondering if there are issues related
to asterisk installation on FC3.
Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails
in my zaptel build trying to find a Makefile in the
/lib/modules/2.6.9/build directory - thanks.
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Richard Lyman wrote:
Brian West wrote:
Symlink /lib/modules/2.6.9/build to /usr/src/linux
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven P. Donegan
Sent: Sunday, November 28, 2004 10:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk
Andy Burns wrote:
Richard Lyman wrote:
Brian West wrote:
Symlink /lib/modules/2.6.9/build to /usr/src/linux
shouldn't that be 'to /usr/src/linux-2.6'
Yes, also FYI I had problems building zaptel 1.0 on 2.6.9-1.681_FC3smp
(error with a reference to non-existent sk_buf-ethernet.mac or
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dave Cotton
Sent: Sunday, November 28, 2004 11:41 AM
To: Asterisk List
Subject: Re: [Asterisk-Users] Asterisk/linux 2.6.9 kernel build failure
On Sun, 2004-11-28 at 08:29 -0800, Steven P. Donegan wrote:
Has anyone successfully built Asterisk
Peter Svensson wrote:
On Sun, 28 Nov 2004, Brian West wrote:
I don't agree with this patch yet... It's the distro's fault for doing this
wrong and I don't feel we have to work around it. The few people I talked
to have Symlinks the build to /usr/src/linux or the like. Then again I
may be
Andrew Thompson wrote:
Christopher Jacob wrote:
All,
This may be a stupid question, but here it is...
What interface gives the most density? Do I top out at T1's? For
instance, 4
t1's to the Digium Quad span t1 card. Is there an interface available
for T3
or DS3?
Depending on where you using
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
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Carmi Weinzweig wrote:
Has anyone compared * to sipXpbx? From a cursory look, this open
source version of PingTel's PBX has many features that make it more
suitable as a replacement for a traditional PBX, including the ability
for users to tell if a phone/trunk is in use. What I am trying to
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy with each
other.
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display
are like yours but without the
callerid= entry and my CS phones give me the received callerid fine.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven P.
Donegan
Sent: 12 September 2004 16:55
To: [EMAIL PROTECTED]; Asterisk Users Mailing List
Firmware now current (1.0.5.11) - no change in what is displayed on the
phone. Good thought though :-)
Steve Maroney wrote:
Try upgrading the firmware
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Steven P. Donegan wrote:
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote
[EMAIL PROTECTED] wrote:
Hi,
I am setting up an Asterisk system with Cisco 7960 phones. I have a
PoE injector to insert between the patch panel and HP 2626 switch. I
plan to plug the users pc into the phone and the phone into the wall.
I would like the phones to have a seperate subnet from
Beierlein Moritz wrote:
want to get from SIP to ISDN or from SIP to SIP.
I only have a ADSL connection that means 786kb/s downstream and 128kb/s
upstream so i can max handle 2 sip calls at once.
I want to have 25 Accounts because of the different numbers for the
different phones.
Good, i wanted
than $50/each. $100 more of it's a P4 instead of a Celeron. Add a
case+PS for $40-$50.
-Original Message-
From: Steven P. Donegan [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 08, 2004 7:45 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System Reqirements HELP
I bought a 5013G
Leif Madsen wrote:
On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote:
For a few years now I've operated with cable as the obvious choice, at least
in my area where RoadRunner really built up a good network. It could be that
for nation wide implementation VoIP really should
Scott Petersen wrote:
On Wed, Aug 04, 2004 at 04:37:32PM -0400, Seth Remington wrote:
On Wed, 2004-08-04 at 14:21, Scott Petersen wrote:
Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.
Maybe I am misunderstanding you but why does
All in favor of IAX with native encrypted tunneling say Aye :-)
Now I'm likely in the target rings of Big Brother :-)
Jay Milk wrote:
Yikes... I don't think it should be too problematic with PSTN
termination, but if you're making VOIP-to-VOIP calls, you will only act
as a SIP Proxy (or somesuch)
Wolfgang S. Rupprecht wrote:
Me raises his hand.
All in favor of IAX with native encrypted tunneling say Aye :-)
Now I'm likely in the target rings of Big Brother :-)
If the voice data passed through a service provider run asterisk
system, I'd imagine they'd just get a court order to
Can Asterisk light the message waiting light on a Grandstream BudgeTone
phone? If so please reply with any related configlets :-)
Thanks!
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To
Chris wrote:
- Original Message -
From: Steven P. Donegan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 01, 2004 10:20 AM
Subject: [Asterisk-Users] Grandstream Message Waiting light
Can Asterisk light the message waiting light on a Grandstream BudgeTone
phone? If so
What is the module name for the TDM40B - I received my X100P and TDM40B today (thanks
Digium).
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is modprobe wcfxs
Andy
*** REPLY SEPARATOR ***
On 27/06/2003 at 16:07 Steven P. Donegan wrote:
What is the module name for the TDM40B - I received my X100P and TDM40B
today (thanks Digium).
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http
I disagree - for many tasks a GUI would be just fine, for others direct coding would
do the trick. They do not have to be mutually
exclusive.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Thursday, June 26, 2003 4:42 PM
To: [EMAIL
it is, is an interface to write out a .conf file
On Thu, 26 Jun 2003 17:04:28 -0700, Steven P. Donegan wrote:
I disagree - for many tasks a GUI would be just fine, for others direct coding would
do the trick. They do not have to be mutually
exclusive.
-Original Message-
From: [EMAIL PROTECTED
a .conf file
On Thu, 26 Jun 2003 17:04:28 -0700, Steven P. Donegan wrote:
I disagree - for many tasks a GUI would be just fine, for others
direct coding would do the trick. They do not have to be mutually
exclusive.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
A BRI has 2 B channels (voice or data at 64k) and 1 D channel (signalling at 16k), a
PRI has 23 B channels and 1 D channel (64k in
this case). From a telephony viewpoint that means a BRI has two voice channels and a
PRI has 23.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Regards,
Michael
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven P. Donegan
The openh323 code is not the 'latest cvs' but the one offered on their
web
site for ftp. Not sure how one would go about getting an 'old' release
I've done this, with the exact versions you state, 3 times today - every one
does the full , proper thing. I did:
cd pwlib;make clean;make opt;make install
cd ../openh323;make clean;make opt;make install
cd ../asterisk/asterisk/channels/h323;make clean;make install;make samples
works every time
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