RE: [Asterisk-Users] Hardware

2004-10-28 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Marcelo Pacheco
 Sent: 26 October 2004 23:23
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Hardware
 
 I have a system with VIA chipsets with one T400P (3FXS,1FXO) 
 and 2 E100P (for testing with a cross over cable between 
 them)

I have an AMD system with Via chipsets, and have problems with an X100P
as I've seen discussed elsewhere on this list.

What I'm trying to establish, is whether the problem lies in the
hardware (ie: the physical card does not get on with the motherboard),
or whether its driver related (ie: the wxfxo driver does not get on with
the system drivers for the Via chipset).

The reason I'm asking, is that I'd like to upgrade to a TDM400 card with
1 x FXO and 1 x FXS modules. Obviously if the problem is hardware
related, then the TDM should be fine, but if software related, I really
don't want (and can't afford) to buy a TDM400 + modules only to find out
it has the same problem!

Does anyone have a TDM400 in an AMD system with a Via chipset,
preferably the one as listed below, that could offer any guidance.
The motherboard is an MSI KM2M Combo-L SKT A KM266 with a Duron 1800.

Any and all help gratefully received,

Cheers,

Karl

bash $ /sbin/lspci 
:00:00.0 Host bridge: VIA Technologies, Inc. VT8375 [KM266/KL266]
Host Bridge
:00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266
AGP]
:00:07.0 Communication controller: Individual Computers - Jens
Schoenfeld Intel 537
:00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB
1.1 Controller (rev 80)
:00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB
1.1 Controller (rev 80)
:00:10.2 USB Controller: VIA Technologies, Inc. VT82x UHCI USB
1.1 Controller (rev 80)
:00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82)
:00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge
:00:11.1 IDE interface: VIA Technologies, Inc.
VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06)
:00:11.5 Multimedia audio controller: VIA Technologies, Inc.
VT8233/A/8235/8237 AC97 Audio Controller (rev 50)
:00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102
[Rhine-II] (rev 74)
:01:00.0 VGA compatible controller: S3 Inc. VT8375 [ProSavage8
KM266/KL266]


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RE: [Asterisk-Users] Almost there--Remote connection

2004-10-19 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


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RE: [Asterisk-Users] Almost there--Remote connection

2004-10-19 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:49
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 Thanks. I think that's Iptables. No?
 I have a hardware firewall.

First, have a peek in rtp.conf and see what it says its using. For
example, my (modified) version looks like:

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=15000
rtpend=17000

 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Deon Rodden
 Sent: Tuesday, October 19, 2004 11:35 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 
 My firewall script has something to the effect of:
 
 # Allow Existing traffic through
 -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
 
 # Incoming VOIP Ports
 -A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 
 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 
 5036:5045 -j ACCEPT -A INPUT -m state --state NEW -m udp -p 
 udp --dport 2727:2727 -j ACCEPT -A INPUT -m state --state NEW 
 -m udp -p udp --dport 4569:4569 -j ACCEPT -A INPUT -m state 
 --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT -A 
 INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j ACCEPT
 
 That's for IAX2 and SIP.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Karl Dyson
 Sent: Tuesday, October 19, 2004 11:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Ferguson, Michael
  Sent: 19 October 2004 16:18
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Almost there--Remote connection
  
 
 [snip]
 
  
  The * server is behind a firewall and I have opened ports 
 1-10100 
  5060
  5004
  4569
  
 
 IIRC, SIP uses 1-2 by default. Have you changed this 
 to 1-10100?
 
 Cheers,
 
 Karl
 
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RE: [Asterisk-Users] Almost there--Remote connection

2004-10-19 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 18:30
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Almost there--Remote connection
 
 I just realised that I neglected to mention that the remote 
 GS100 phone is sitting behind a firewall also.
 Do I need to open any outgoing ports on that firewall? 
 Considering that one cannot hear anything from the GS100 IP phone?
 

Yes, both phones will need to have ports 1-2 open (having seen
your rtp.conf) if they are going o register with your * server.
 
 
 Mine says
 rtpstart=1
 rtpend=2
 



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RE: [Asterisk-Users] X100P red alert

2004-10-19 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Kristian Kielhofner
 Sent: 19 October 2004 22:12
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] X100P red alert
 
 Alex van Es wrote:
 
  Hi all,
  I just got my X100P card installed and asterisk keeps on 
 complaining 
  that it cannot create a zap channel. I read somewhere on 
 the internet 
  that the zap card will not work when a phoneline is not plugged in, 
  cause is draws power from the phoneline. Is this correct? Of course 
  eventually I will connect it to a phoneline, but I would 
 just like to 
  know this for sure..
  
  Alex
  
 
 Alex,
 
   It may not be able to create a zap channel because the 
 driver isn't loaded.  Make sure that you have the kernel 
 modules installed and everything working, etc.  Too much to 
 go over right now.  Look in the wiki, etc.  But you will need 
 to have it connected to the phoneline to make calls over the 
 phone line, of course.
 
   The X100p does not get power over the phone line.  The 
 PCI bus takes care of that.
 

Whilst the card does get power from the line, it does detect if the line
is present. If you have the zaptel module loaded, and the ztcfg runs ok,
you should have:

bash $ cat /proc/zaptel/1 
Span 1: WCFXO/0 Wildcard X101P Board 1 

   1 WCFXO/0/0 FXSKS (In use) 

This will say RED ALARM if the card is unable to use the line. Check the
cord you have plugged in is the correct cord, and has the correct pins
connected. I had this originally, and on swapping the cord all worked
fine :)

Also:

bash $ ls -al /dev/zap/*
lr-xr-xr-x  1 root root7 Jan  1  1970 /dev/zap/1 - span1/1
crw-rw-rw-  1 root root 196, 254 Jan  1  1970 /dev/zap/channel
crw-rw-rw-  1 root root 196,   0 Jan  1  1970 /dev/zap/ctl
crw-rw-rw-  1 root root 196, 255 Jan  1  1970 /dev/zap/pseudo
crw-rw-rw-  1 root root 196, 253 Jan  1  1970 /dev/zap/timer

/dev/zap/span1:
total 0
drwxr-xr-x  1 root root  0 Jan  1  1970 .
drwxr-xr-x  1 root root  0 Jan  1  1970 ..
crw-rw-rw-  1 root root 196, 1 Jan  1  1970 1
bash $ 

Hope this helps,

Cheers,

Karl


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RE: [Asterisk-Users] Intel Modem vs Digium Cards

2004-10-12 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Adam Goryachev
 Sent: 12 October 2004 04:20
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards
 
  Wow, that's a really sucky attitude.  I would expect 
 *Digium* to tell 
  him to go away and solve his own problems.  However, if the user 
  community does that, then this is one of the suckiest user 
 communities 
  I've run across in the free software world, and I've been 
 doing free software for many years.
 
 Well, IMHO, I would expect it perfectly reasonable for one of 
 three responses from the 'user' community (in order of likelihood):
 
 1) A resounding non-response
 2) A response of Well, get a X101P or TDMx0P and try it
 3) Some advice on how to resolve the problem.
 

I think it's a good thing the rest of the user community (IMHO) doesn't appear to 
follow your lead then!

I bought a £20 generic X101P from a local voip hardware supplier, and am rather 
pleased I did. It works most of the time, but drops calls occasionally. I originally 
thought this might be due to it being a cheap card, but I now find out that the Digium 
X100P/X101P is the same card with a heatsink, and is 5 times more expensive. Nice. 
Additionally, the problem appears to be that the AMD Duron/VIA chipset combination in 
my server doesn't get along with the X10(0|1)P cards, and so if I'd bought the pricey 
card, I'd still have problems, only solvable by another purchase, and also be 
significantly out of pocket.

FYI: I found all this out by use of Google, and the user community. I go searching for 
my solutions, and have not called Digium once, and would not expect them to answer 
calls on my card.

To resolve my problem, I may buy a Sipura-3000, I may buy a Digium TDM card. One I 
woiuld expect support from Digium for, one I would not.

Both I would expect to be able to *ask* a question here, and I would *hope* that 
people that had heard of what I'm using would respond. I would also hope that people 
that had ever heard of it would keep quiet rather than waste my bandwidth baiting me 
as to why I didn't buy their pet hardware choice of the moment.

Cheers for now,

Karl


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RE: [Asterisk-Users] Intel Modem vs Digium Cards

2004-10-12 Thread Karl Dyson
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Benjamin on Asterisk Mailing Lists
 Sent: 12 October 2004 09:36
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards
 
 On Tue, 12 Oct 2004 09:05:02 +0100, Karl Dyson [EMAIL PROTECTED] wrote:
  It works most of the time, but drops calls occasionally.
 
 and:
 
  To resolve my problem, I may buy a Sipura-3000
 
 I don't know how much effort you have already made but keep 
 in mind that there are parameters in the Zaptel drivers which 
 can be tweaked to at least reduce false hangups significantly.

It doesn't look like false hangups (IMHO). Calls drop mid call,
regardles of volume. You can go for a week with no drops, and then get 5
in a day (with fairly consistent usage patterns).

Don't worry about effort involved -- it's installed in my house, and I
tinker when ever possible, and enjoy it. My wife isn't especially over
the moon with the dropped calls, but is a patient woman :)

 There is also a patch now which allows you to disable the 
 disconnect supervision on outgoing Zaptel calls only because 
 that's the scenario where you are most unlikely to need it.
 

I get dropped calls occasionally on both in and outbound calls, and to
aid diagnosis have a phone plugged directly into the line as well now (a
splitter now has the connection to the X100P and this additional
handset). If an inbound call drops, you can pick up this handset and
continue the call, this indicating the X100P is dropping the call.

I'd quite like to keep the internal card, and hence like the idea of
upgrading to a TDM400 This would allow me to continue to use
distinctive ringing, especially as I have a use for it(!), but I don't
want to spend money on a TDM11B kit only to find that it doesn't get
along with the amd/via combination. Is anyone using a TDM400 of any
description (preferably FXO *and* FXS just in case) with an AMD/Via
combination?

I don't think the Sipura does DRING, so would have to loose that
functionality if I go that route. On the plus side, I get failover
FXO-FXS on the sipura in the event the * server dies, or in the event
of a power failure. Swings and roundabouts!

Thanks,

Karl


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RE: [Asterisk-Users] Zaptel 1.0.0. will not compile

2004-10-10 Thread Karl Dyson
Personally I get zaptel, zapata etc from cvs rather than portage.

Check your /usr/src/linux symlink points to the correct place... I got
all sorts of grief with gentoo when I forgot to put it back after some
playing.

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Remco Barende
 Sent: 10 October 2004 21:48
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Zaptel 1.0.0. will not compile
 
 OK, I got a little further. I don't know why but after 
 re-emerging zapata finally zaptel will build. It is not 
 working however and I still get too many erros during the 
 build. It's now complaining about:
 
CC [M]  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.o
 /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:163: 
 warning: 
 `fcstab' defined but not used
CC [M]  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/tor2.o
CC [M]  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/torisa.o
 /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/torisa.c:1139:
  warning: 
 `set_tor_base' defined but not used
 
 
 *** Warning: zt_register 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_transmit 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_receive 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_ec_chunk 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_set_dynamic_ioctl 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_unregister 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_alarm_notify 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_rbsbits 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdynamic.ko]
  has no CRC!
 *** Warning: zt_transmit 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] 
 has no CRC!
 *** Warning: zt_receive 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] 
 has no CRC!
 *** Warning: zt_unregister 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] 
 has no CRC!
 *** Warning: zt_register 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztdummy.ko] 
 has no CRC!
 *** Warning: zt_dynamic_unregister 
 [/var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/ztd-eth.ko] 
 has no CRC! 
 and many many more
 
 
 
 On Sun, 10 Oct 2004, Remco Barende wrote:
 
  Hi list
 
  I am trying to install asterisk on a gentoo box running 
 kernel version 
  Linux version 2.6.8-gentoo-r7 (gcc version 3.3.4 20040623 (Gentoo 
  Linux 3.3.4-r1, ssp-3.3.2-2, pie-8.7.6)) #1 Thu Oct 7 20:24:31 CEST 
  2004
 
  I tried to install from the provided ebuild for Asterisk 
 1.0.0 but the 
  compile of the zaptel module fails miserably.
 
  Compilation seems to start at first but then generates 
 pages and pages 
  of errors like :
 
  include/linux/types.h:18: error: syntax error before 
 __kernel_dev_t
  include/linux/types.h:18: warning: type defaults to `int' in 
  declaration of `__kernel_dev_t'
  include/linux/types.h:18: warning: data definition has no type or 
  storage class
  include/linux/types.h:21: error: syntax error before dev_t
 
  include/linux/types.h:152: warning: type defaults to `int' in 
  declaration of `f_tinode'
  include/linux/types.h:152: warning: data definition has no type or 
  storage class
  include/linux/types.h:155: error: syntax error before '}' token In 
  file included from
  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:42:
  include/linux/kernel.h:15:27: asm/byteorder.h: No such file or 
  directory
  include/linux/kernel.h:16:21: asm/bug.h: No such file or 
 directory In 
  file included from
  /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.c:42:
  include/linux/kernel.h:81: error: syntax error before size_t
  include/linux/kernel.h:82: warning: function declaration isn't a 
  prototype
  include/linux/kernel.h:82: warning: conflicting types for built-in 
  function `snprintf'
  include/linux/kernel.h:83: error: syntax error before size_t
  include/linux/kernel.h:83: warning: function declaration isn't a 
  prototype
 
  include/linux/time.h:145:31: division by zero in #if
  include/linux/time.h:145:31: division by zero in #if
  include/linux/time.h:145:31: division by zero in #if
  include/linux/time.h:145:31: division by zero in #if
 
  include/linux/fs.h:356: error: storage size of `bd_sem' isn't known
  include/linux/fs.h:357: error: storage size of `bd_mount_sem' isn't 
  known
  include/linux/fs.h:431: error: storage size of `i_atime' isn't known
  include/linux/fs.h:432: error: storage size of `i_mtime' isn't known
 
 
  Ideas anyone?
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RE: [Asterisk-Users] Asterisk 1.0 released

2004-09-23 Thread Karl Dyson
Can anyone confirm if the UK callerid patches were incorporated into CVS
or this release? I am still using an older version with the patches
applied, and they are working fine, but I cannot give up this
functionality.

Thanks in advance,

Karl 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol
Sent: 23 September 2004 15:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk 1.0 released

Hi,

Reporting from Astricon, Mark uploaded the 1.0 release while giving his
speech a few mintues ago..

Bring out the champagne :)

Lethol
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RE: [Asterisk-Users] Digium X100P vs Dodgy Ebay X100P

2004-06-16 Thread Karl Dyson
I wouldn't mind a look at the eBay one, perhaps at a smaller res though
(1024x768 ?)

I've got one of them waiting to be fitted, and I'd be interested to see
if it's the same one.

--
-S

Me too. I'm using a clone X101P I  bought from Goods2World a while back
and get occasional dropped calls, so would be interested in comparing
the two images. If you could stick them on a web server they could be
downloaded easily enough (personally I'd prefer the full images).

Cheers,

Karl


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RE: [Asterisk-Users] How to get the Called id with AGI

2004-06-11 Thread Karl Dyson
Hi,

I lookup incoming calls in a MySQL DB to try and match up names and
display them with the number on the 7905s. Inbound pstn calls land based
on dring settings, so I pass (in this case 918) to be added to the
display name to indicate which of my inbound numbers was dialled. So, on
the 7905 I get

918:Callers Name
01234567890

Or,

918:Unknown
01234567890

if I don't recognise the number.

Hope this helps,

Karl

### extensions.conf ###

[inbound-pstn-1]
exten = s/,1,NoOp(${EXTEN})
exten = s/,2,Goto(inbound-pstn-nocli,s,1)

exten = s,1,NoOp(DRING:918)
exten = s,2,AGI(clilookup.pl,918)
exten = s,3,Goto(inbound-pstn,s,3)

exten = t,1,Hangup

exten = h,1,Hangup

### clilookup.pl ###

#!/usr/bin/perl
#

my $debug = 1;

use Asterisk::AGI;
use DBI;

$AGI = new Asterisk::AGI;

my $dsn = 'dbi:mysql:database=asterisk;host=snip';
my $dbuser = 'snip';
my $dbpass = 'snip';

my $dbh = DBI-connect($dsn, $dbuser, $dbpass,{ PrintError = 0,
RaiseError = 0, AutoCommit = 0 });

my %input = $AGI-ReadParse();

foreach $thing (sort keys %input) {
   $res = $input{$thing};
   $AGI-verbose(input{$thing} = $res,1) if $debug;
}

my $id = shift;
$id .= : if $id;
$AGI-verbose(id = $id,1) if $debug;

$cli = $input{callerid};
my $qcli = $dbh-quote($cli);

my $sth = $dbh-prepare(qq{select name from numbers where number=$qcli})
|| die;

$sth-execute || die;

my $cliname;
if($sth-rows  0)
{
  $cliname = $sth-fetchrow_array;
}
else
{
  $cliname = Unknown;
}
$sth-finish;

$AGI-set_callerid(\${id}${cliname}\$cli);

$dbh-disconnect;
exit 0;

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of usedcanon
 Sent: 10 June 2004 23:33
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] How to get the Called id with AGI
 
 Hi Angel,
 
 I assume you mean CALLERIDNUM (the number part of caller ID),
 
 The easiest thing in that case is to pass it as a parameter to your
AGI
 script
 
   extex = 500,1,AGI(myscript.py|${CALLERIDNUM})
 
 in your script you just used the argument passed as usual ( I am not a
 perl expert, so not sure on the syntax there )
 
 Hope this helps.
 
 Umar.
 
   -Original Message-
   From: [EMAIL PROTECTED]
[mailto:asterisk-users-
 [EMAIL PROTECTED] Behalf Of Angel Diaz
   Sent: 10 June 2004 22:41
   To: [EMAIL PROTECTED]
   Subject: [Asterisk-Users] How to get the Called id with AGI
 
 
   Hi all,
   Is there a way to get the called id (the B number) with
AGI
 perl ?
   I know how to get the caller id which is working fine and is
just
 below:
   #!/usr/bin/perl
 
   use Asterisk::AGI;
 
   $AGI = new Asterisk::AGI;
 
   my %input = $AGI-ReadParse();
 
   $callerid = $input{'callerid'};
 
   $AGI-say_digits($callerid);
 
   }
 
   Thanks in advance,
 
   Angel.
 
 
 
 
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 http://messenger.yahoo.com/
 
 


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RE: [Asterisk-Users] How to get the Called id with AGI

2004-06-11 Thread Karl Dyson
snip

Just realised I answered completely the wrong question. Misread
called id as caller id.

D'Oh.

Sorry,

Karl



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RE: [Asterisk-Users] re: Voicemail and Cisco Phones

2004-06-07 Thread Karl Dyson
On my 7905s I can configure a voicemail number, which in turn
activates a Messages softkey. When pressed, it goes to that exten,
which in turn is configured to go to VoiceMailMain. Great, works like a
charm. However, the phones also have a go to voicemail timeout after
which, the phone diverts the call to voicemail via a temporarily
moved. This doesn't work, as it diverts the call to the VoiceMailMain
exten, and so instead of taking a message, prompts for a password.
So I currently have the timeout on the phone set higher than the
timeout on the * box, and so * bounces the call to voicemail (correctly)
after the required timeout.
I'd like to get the feature working via the phones though, as the To
VM softkey and DND features do not work corectly as they function in
the same way (it seems).

Does anyone know if this is something I can fix?

Cheers,

Karl

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt
Sent: 07 June 2004 13:52
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] re: Voicemail and Cisco Phones


The Cisco 7960 has a softkey called DND which when pressed as the phone
is ringing will sack the call to voicemail.  If you where using Cisco
CME or CM you can forward all calls to Vmail via CLI or GUI.

Kurt




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RE: [Asterisk-Users] Re: Caller ID with BT CD50

2004-05-31 Thread Karl Dyson
Ok, I can report that I have just (within the last 2 hours, anyway)
downloaded the current cvs head for zaptel and asterisk, and applied
Tony's current patches downloaded freshly this evening from nodomain.
All applied and compiled, and with a tweak to my dring statements after
running asterisk in debug, cli and dring are both working :)

Calls with no CLI on either number go straight to voicemail without
ringing a phone (usually cold callers in my experience), and I lookup
the caller id in a mysql db using an AGI written in perl, that also
allows me to display which number the caller dialled as well as a
friendly name against the cli in the 7905 displays :)

Thanks All,

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Karl Dyson
 Sent: 30 May 2004 00:33
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Re: Caller ID with BT CD50
 
 Certainly Tony's original patch for CID works with my generic X101P
 (reports itself as an Intel 537 IIRC). I will get around to
downloading
 his new single patch that includes distinctive ringing and testing it
in
 the next couple of days.
 
 Cheers for now,
 
 Karl
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Tony Mountifield
  Sent: 29 May 2004 19:06
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Re: Caller ID with BT CD50
 
  In article [EMAIL PROTECTED],
  Tony Hoyle [EMAIL PROTECTED] wrote:
   Kevin Walsh wrote:
  
I downloaded the latest version of your patch, from your
website,
 and
it works perfectly.  I had waited until I had some time
available
because I thought I'd have to play around with it for a while.
  
   Great.   Just need to make sure that it still works for US lines
and
  it's all set.
  
   There's some debate whether to use this patch or to wait for one
 that
  uses
   line reversal/guard tone detection...   there is the slight
problem
 that
  the
   X100P can't detect line reversal so it'd mean everyone upgrading
 their
   hardware...  still, I have what 'works for me' and will continue
 hosting
  it
   for a while whatever happens.
 
  Is that the X100P generically, including the X101P?
 
  I would include both algorithms - the line reversal one for the
 hardware
  that can do it, and your current one for those that can't.
 
  Cheers,
  Tony
  --
  Tony Mountifield
  Work: [EMAIL PROTECTED] - http://www.softins.co.uk
  Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-28 Thread Karl Dyson
Just tried to apply the patch:

Just checked out asterisk stable and zaptel, patched using Tony's
patches (which worked, and compiled previously)

Then got this when applying your patch.

bash # cat ../chan_zap.c.diff | patch -p0
patching file channels/chan_zap.c
Hunk #1 succeeded at 4642 (offset -148 lines).
Hunk #2 FAILED at 4681.
1 out of 2 hunks FAILED -- saving rejects to file
channels/chan_zap.c.rej
bash #

Cheers,

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Kevin Walsh
 Sent: 28 May 2004 09:12
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Caller ID with BT CD50
 
 Robert Boardman [EMAIL PROTECTED] wrote:
  First of all thanks for the patch it works great,
 
  but i think it breaks the distinctive ringing,
  I have 2 incoming numbers in one x100p in contexts home1 and home2
but
  'default' is always chosen has anyone else seen this?
 
 Yes - it does break the distinctive ring detection, but that's easily
 sorted out.
 
 The correct way would be to move the if (p-use_callerid == 2)
 code within the existing if (p-use_callerid) block, with a couple
 more if conditionals here and there.  The quick way, however, is
 to apply the attached chan_zap.c hack over the top of Tony Hoyle's
 great work.
 
 In the standard chan_zap.c, you can't have distinctive ring detection
 unless you also need Caller*ID detection.
 
 My hack makes two changes:
 
 1. Changes an else if into an if to get the world = USA
Caller*ID code to run.  This will waste a little time, but
no more than we were wasting anyway, before Tony's patch was
applied.
 
 2. Comment out a line of code to ensure that we always answer
after the first ring.  We need the first ring to give the
the distinctive ring code something to work with, of course.
 
 It works for me.  Hopefully it'll work for you too.
 
 --
_/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
   _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s
h
  _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
 _/   _/  _/_/_/_/  _/_/_/_/  _/_/
 
 


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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-28 Thread Karl Dyson
Oddly, it looks like the changes were made(!?)

It might be, having read Tony's reply, that it's because I applied the
uk cli patches from Tony and yourself to the stable rather than head
branches?

I'll try compiling and let you know.

Cheers for now,

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Kevin Walsh
 Sent: 28 May 2004 12:07
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Caller ID with BT CD50
 
 Karl Dyson [EMAIL PROTECTED] wrote:
  Just checked out asterisk stable and zaptel, patched using Tony's
  patches (which worked, and compiled previously)
 
  Then got this when applying your patch.
 
  bash # cat ../chan_zap.c.diff | patch -p0
  patching file channels/chan_zap.c
  Hunk #1 succeeded at 4642 (offset -148 lines).
  Hunk #2 FAILED at 4681.
  1 out of 2 hunks FAILED -- saving rejects to file
  channels/chan_zap.c.rej
  bash #
 
 Could you try applying the changes by hand.  There are only two lines
 to change and it looks as if the first one went through.  I'll check
 my patch to see if I messed up the original or something silly.
 
 --
_/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
   _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s
h
  _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
 _/   _/  _/_/_/_/  _/_/_/_/  _/_/
 
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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-28 Thread Karl Dyson
Well compiles and runs OK, but it doesn't identify the dring. I only
started playing with it this morning (only realised it *did* dring when
I saw your it's broken dring post)

This is what I have in zapata.conf

dring1=95,0,0
dring1context=inbound-pstn-1
dring2=325,95,0
dring2context=inbound-pstn-2

is this correct for the UK? (I suspect not, and yes, I have dring on my
bt line).

Cheers,

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Karl Dyson
 Sent: 28 May 2004 12:36
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Caller ID with BT CD50
 
 Oddly, it looks like the changes were made(!?)
 
 It might be, having read Tony's reply, that it's because I applied the
 uk cli patches from Tony and yourself to the stable rather than head
 branches?
 
 I'll try compiling and let you know.
 
 Cheers for now,
 
 Karl
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Kevin Walsh
  Sent: 28 May 2004 12:07
  To: [EMAIL PROTECTED]
  Subject: RE: [Asterisk-Users] Caller ID with BT CD50
 
  Karl Dyson [EMAIL PROTECTED] wrote:
   Just checked out asterisk stable and zaptel, patched using Tony's
   patches (which worked, and compiled previously)
  
   Then got this when applying your patch.
  
   bash # cat ../chan_zap.c.diff | patch -p0
   patching file channels/chan_zap.c
   Hunk #1 succeeded at 4642 (offset -148 lines).
   Hunk #2 FAILED at 4681.
   1 out of 2 hunks FAILED -- saving rejects to file
   channels/chan_zap.c.rej
   bash #
  
  Could you try applying the changes by hand.  There are only two
lines
  to change and it looks as if the first one went through.  I'll check
  my patch to see if I messed up the original or something silly.
 
  --
 _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
_/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l
s
 h
   _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
  _/   _/  _/_/_/_/  _/_/_/_/  _/_/
 
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RE: [Asterisk-Users] Conference Server

2004-05-27 Thread Karl Dyson
If you have usb hardware installed, you can use the ztdummy driver (part
of the zaptel bits), and you don't need usb hardware if you're using a
2.6 kernel IIRC

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of pesb
 Sent: 27 May 2004 16:59
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Conference Server
 
 Hi there,
  I need to implement a SIP Conference Server. I've saw
that
 asterisk has an application called meetme. But, it says that A ZAPTEL
 INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.
 Is there any other way to implement a conference server without the
need
 of
 having a ZAPTEL Interface?
 I need my conference server to work only with my SIP Phones.
 
 thanks in advance,
   Pablo Salinas
 
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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-26 Thread Karl Dyson
Can confirm it works with Generic X101P

*BIG* Thank you :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh
Sent: 26 May 2004 16:15
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Caller ID with BT CD50

Tony Hoyle [EMAIL PROTECTED] wrote:
  Lost some of the mails on this topic somewhere.
  
  Does this need the BT50 mod or will the X100p now output the Caller
ID?
  
 It's to allow the X100P to output the caller ID.
 
I can confirm that it works for me.

I applied the patches, compiled and installed zaptel, compiled and
installed Asterisk and it just worked.

Well done.  I'll abandon my attempt now. :-)

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk firewall config

2004-05-24 Thread Karl Dyson
Ah yes. I too would like to see ip_conntrack_sip :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Stenton
Sent: 24 May 2004 08:57
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk firewall config

If your firewall has some form of sip inspect then you will not need to
leave open the rtp ports.

Chris

- Original Message -
From: Tony Hoyle [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 11:11 PM
Subject: [Asterisk-Users] Asterisk firewall config


 The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to
the
 world to work.  Is this necessarily true, or does it only need some of
these
 outgoing?

 I'm concerned as anyone that could guess an extension numberpassword
could
 use my server to make outgoing calls.  It would help if the extensions
had
a
 netmask/allowable IP setting like the iax.conf file uses, but there
isn't
one
 documented...

 Tony

 -- 
 Te audire no possum. Musa sapientum fixa est in aure.

 Tony Hoyle [EMAIL PROTECTED]  Key ID: 104D/4F4B6917 2003-09-13
 Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-23 Thread Karl Dyson
Well, I have a USR015630B, which, according to the FAQ supports (UK)
CLI. It supports the at #cli command, but no matter what I try, it will
not pick up the caller id. Lucky I already had it and didn't buy it
soley for this purpose! My caller display unit (unfortunately a CD60 --
which I've opened to look for CD50 like boards, chips etc) picks up CLI
no problem (as you'd expect), as do my DECT phones. of course, only
when they're plugged into the line, and not when they're plugged into
the ATA186.

I will be persuading someone I know to swap the USR for a Hayes modem
later today to see if that will do it. otherwise out will come the
soldering iron, and eBay will see me bidding on CD50s!

Cheers,

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Tony Hoyle
 Sent: 22 May 2004 23:55
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Caller ID with BT CD50
 
 Karl Dyson wrote:
 
  Hi All,
 
  Having searched the archives, I can see there has been much
discussion
  at various points regarding capture of caller id information from
good
  old BT.
 
  If I understand correctly, it seems that not only do the drivers not
  currently support it, but my X101P possibly/probably can't do it
anyway
  due to hardware?
 
  From the details on http://www.ainslie.org.uk/callerid/cli_faq.htm it
 sounds
 like it wouldn't be too hard to implement, however:
 
 The only manufacturers that have ever supported BT Caller ID are
Pace,
 Hayes
 (Europe), and 3Com/US Robotics.
 
 It then goes on to state all 3 of those manufactures no longer support
it.
 
 I wonder if the low cost geographic VOIP numbers support it?
 
 Tony
 
 --
 Te audire no possum. Musa sapientum fixa est in aure.
 
 Tony Hoyle [EMAIL PROTECTED]  Key ID: 104D/4F4B6917 2003-09-13
 Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-23 Thread Karl Dyson
 I have a USR modem too but it's a brick unfortunately...  Pity as it
was
 a nice modem... the right model too so it might have worked.
 
 I wonder what it would take to get the zaptel drivers to pick up CID
(or
 even a cheap conexxant modem or something like that) - these soft
modems
 are pretty dumb so I'd expect there to be some scope for frobbing them
 to do new stuff.
 
 Tony
 

As I say, I lent a Hayes modem to mother in law a year ago, but have
since upgraded her to a 56k internal, but cunningly left the old kit in
her loft (think of it as redundant storage of my old computer crap ;))
so will pick it up later and try it. failing that, I've located a
possible Pace candidate

Of course, although my wife is happy with the Cisco 7905s that have
sprung up around the house, she still likes the cordless DECT units we
have, and so they're plugged into an ATA186. Problem is, they no longer
display caller id due to the ATA186 not poking it out in BT format I
guess. If I were to buy some US cordless handsets would they do the
caller id display? Or am I pushing my luck now! (I'm afraid, nice as
they look, the Cisco 7920 cordless phones are a bit out of my price
range!!)

Cheers,

Karl



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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-23 Thread Karl Dyson
 
 ATA186 firmware 3.0+ supports more formats.
 
 Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine.

Oooh Now which setting would I need to check?? I have a Philips Onis
DECT system, which does CLI quite happily on the BT line, and I just
checked the ATA and its running :

ata000e841adb36
Version: v3.0.0 atasip (Build 031210A)
MAC: 0.14.132.26.219.54
SerialNumber: INM07491B0E
ProductId: ATA186I2
Features: 0x
HardwareVersion: 0x0006 0x

So. I guess I must have a setting incorrect??

Thanks in advance,

Karl



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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-23 Thread Karl Dyson
 If you had CID to start with, I'd expect it to work - eg. you wouldn't
get
 it
 from the POTS line but if a VOIP call came in and the ATA186
retransmitted
 that in US format then a US handset would pick it up.

But once I can get the cid working, I'm hoping I can persuade the DECT
units to pick it up obviously the 7905s display CID happily. but
currently only from each other, or incoming IAX calls.
 
I always wondered why phones tended to blip a second before
 starting to ring these days.. now I know).

Me too. suddenly dawned on me recently that old phones would blip
due to the CLI coming through moments before the ring



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RE: [Asterisk-Users] Caller ID with BT CD50

2004-05-23 Thread Karl Dyson
 

http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administr
at
 ion_guide_chapter09186a00801e0eb0.html#wp1113416
 
 ETSI method (type 2).
 
 I know my CD50 still doesn't like this method, but then as the first
 generation device it can be *very* fussy at the best of times -
indeed, I
 think very short line lengths were a problem with this device, they
have
 problems with ISDN where the local NTE actually generates CDS.
 
 But as said the BT DECT 3010 is fine, and so is my CD20 etc.

Thanks for that I had only really thought of it while typing one of
my earlier replies about inbound CLI, and has automatically assumed that
the ata186 would not do CLI in a BT/UK way!

It looks like the Philips units don't like it anyway (they are ~4 yrs
old, mind). I've tried 19e62 and 19e66 so far (the main bits should be
fine as defaults I think, and so I just tried getting it to send it
before ringing, and after the 1st ring).

I'll try the BT CD 60 unit I have in a while to see if that can see the
CLI coming from the ATA.

Cheers,

Karl



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[Asterisk-Users] Caller ID with BT CD50

2004-05-22 Thread Karl Dyson
Hi All,

Having searched the archives, I can see there has been much discussion
at various points regarding capture of caller id information from good
old BT. 

If I understand correctly, it seems that not only do the drivers not
currently support it, but my X101P possibly/probably can't do it anyway
due to hardware?
So, that leaves me with the modem route, which seems more and more
unlikely, due to the seeming difficulties finding a modem that will
*definitely* do it, or the CD50 mod.
Which brings me to my question (finally) Has anyone done this [the
CD50] mod? It seems the CD50 can be found for a few quid, and I'm not
afraid of my soldering iron...
I just wondered how people in the UK were capturing callerid. there
is so much more you can get asterisk to do if you have access to
callerid info.

By the way. much as I'd like to do this by switching to ISDN, and be
done with it, this server is at my home for me to play with, and ISDN is
*not* cheap in fact it would roughly double my quarterly phone
bill. other than the price, ISDN would be my solution of choice!

Thanks in advance,

Karl



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RE: [Asterisk-Users] Asterisk firewall config

2004-05-22 Thread Karl Dyson
I personally only allow IAX2 in and out from my asterisk box, due to the
simplicity of one (udp) port. I do not relish the thought of trying to
open the port ranges for SIP securely!

As long as your inbound stuff in iax.conf lands in a sensible context,
inbound connections would only be able to call your internal extensions,
and not make cost calls.

Hope that helps

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Tony Hoyle
 Sent: 22 May 2004 23:11
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Asterisk firewall config
 
 The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to
the
 world to work.  Is this necessarily true, or does it only need some of
 these
 outgoing?
 
 I'm concerned as anyone that could guess an extension numberpassword
 could
 use my server to make outgoing calls.  It would help if the extensions
had
 a
 netmask/allowable IP setting like the iax.conf file uses, but there
isn't
 one
 documented...
 
 Tony
 
 --
 Te audire no possum. Musa sapientum fixa est in aure.
 
 Tony Hoyle [EMAIL PROTECTED]  Key ID: 104D/4F4B6917 2003-09-13
 Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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