Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

2017-06-05 Thread Marcelo Terres
Try tcpdump

On 5 Jun 2017 9:41 pm,  wrote:

Doesn't matter how much I increase the verbose output
asterisk -vvr
asterisk will not even print a single line.

How to find out if my firewall has this port open?
https://www.grc.com
is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/


Thelma
On 06/05/2017 02:19 PM, Victor Villarreal wrote:
> I think you need to increase verbose output and search in
> /var/log/asterisk/full for any error message related to IAX2 registration
> or simil.
>
> 2017-06-05 17:12 GMT-03:00 :
>
>> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
>> while and it was zoiper was working OK with my previous version of
>> asterisk.
>>
>> After upgrade to 11.25.1 it stop working.
>> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
>> 443 open.
>>
>>
>> Thelma
>> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
>>> Another might be to make sure iptables isn't blocking the connection.
>>>
>>> You can run
>>> iptables -L -n -v
>>> To see if its set to block any ports.
>>>
>>>
>>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
 I'm getting:
 netstat -a |grep 4569
 udp0  0 0.0.0.0:45690.0.0.0:*

 Should I be getting localhost IP?

 Thelma

 On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> Does asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have a problem registering
> Zoiper (IAX) to Asterisk.
> I'm getting an error:
> Registration refused
>

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Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

2017-06-05 Thread Victor Villarreal
Ok..

Maybe you can try with this command, from the Asterisk cli:

cli> core set verbose 5
cli> core set debug 5
cli> iax2 set debug on

And then, try to register with your softphone.

At the end of the test, execute:

cli> iax2 set debug off

And finally, review the logfiles.

2017-06-05 17:40 GMT-03:00 :

> Doesn't matter how much I increase the verbose output
> asterisk -vvr
> asterisk will not even print a single line.
>
> How to find out if my firewall has this port open?
> https://www.grc.com
> is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/
>
>
> Thelma
> On 06/05/2017 02:19 PM, Victor Villarreal wrote:
> > I think you need to increase verbose output and search in
> > /var/log/asterisk/full for any error message related to IAX2 registration
> > or simil.
> >
> > 2017-06-05 17:12 GMT-03:00 :
> >
> >> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
> >> while and it was zoiper was working OK with my previous version of
> >> asterisk.
> >>
> >> After upgrade to 11.25.1 it stop working.
> >> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
> >> 443 open.
> >>
> >>
> >> Thelma
> >> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
> >>> Another might be to make sure iptables isn't blocking the connection.
> >>>
> >>> You can run
> >>> iptables -L -n -v
> >>> To see if its set to block any ports.
> >>>
> >>>
> >>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
>  I'm getting:
>  netstat -a |grep 4569
>  udp0  0 0.0.0.0:45690.0.0.0:*
> 
>  Should I be getting localhost IP?
> 
>  Thelma
> 
>  On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> > Does asterisk listen on port 4569 by default?
> >
> > I'm running version Asterisk 11.25.1 and have a problem registering
> > Zoiper (IAX) to Asterisk.
> > I'm getting an error:
> > Registration refused
> >
> 
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> >>>
> >>>
> >>>
> >>
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> >> org/
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> >
> >
> >
> >
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Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

2017-06-05 Thread thelma
Doesn't matter how much I increase the verbose output
asterisk -vvr
asterisk will not even print a single line.

How to find out if my firewall has this port open?
https://www.grc.com
is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/


Thelma
On 06/05/2017 02:19 PM, Victor Villarreal wrote:
> I think you need to increase verbose output and search in
> /var/log/asterisk/full for any error message related to IAX2 registration
> or simil.
> 
> 2017-06-05 17:12 GMT-03:00 :
> 
>> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
>> while and it was zoiper was working OK with my previous version of
>> asterisk.
>>
>> After upgrade to 11.25.1 it stop working.
>> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
>> 443 open.
>>
>>
>> Thelma
>> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
>>> Another might be to make sure iptables isn't blocking the connection.
>>>
>>> You can run
>>> iptables -L -n -v
>>> To see if its set to block any ports.
>>>
>>>
>>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
 I'm getting:
 netstat -a |grep 4569
 udp0  0 0.0.0.0:45690.0.0.0:*

 Should I be getting localhost IP?

 Thelma

 On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> Does asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have a problem registering
> Zoiper (IAX) to Asterisk.
> I'm getting an error:
> Registration refused
>

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>>>
>>>
>>>
>>
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>> org/
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> 
> 
> 
> 
> 

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Re: [asterisk-users] [SPAM] - Re: queue moh - Email found in subject

2013-07-12 Thread Andrew Thomas
Hi Ioan,

I have done that [Set(CHANNEL(musicclass)=…] but it still doesn’t work when a 
‘queue’ call is put on hold.  So, I can get it to play the correct moh when I 
use the ‘r’ option – but still get silence when I don’t include the ‘r’.

So I’ve sort of fixed my second point (with the ‘r’) but the first point 
(without the ‘r’) is still not working.

Thanks for your feedback ☺

Cheers
Andy


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias
Sent: 10 July 2013 20:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM] - Re: [asterisk-users] queue moh - Email found in subject

Hello Andy,

Have you tried using SetMusicOnHold command before Queue command?

BR,
Ioan

On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote:
Hi All,

Sorry if this has been covered already, but I don't tend to follow this
list as close as I should these days.

Problem is that if a call comes in to a queue without option 'r'
specified - moh plays as expected.  Now, when that call is answered, all
is fine. Trouble comes when that person then puts the caller on-hold.
No moh is heard by the caller (in fact, they get silence).

If I use 'r' - then ringing is heard - but the queue's
musiconhold/musicclass is ignored completely.  When the caller is put on
hold, they do hear moh but the default moh context is used - not the moh
of the queue.

What I need is for the queue's moh to be used when the caller is put on
hold (and without using the 'r' feature).  Is this possible?

* 1.8.16.0 (tried on various flavours of 1.8).
Queue static and realtime (same outcome).

Cheers
Andy












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Re: [asterisk-users] [Spam] Re: Calender and EWS with shared calenders

2013-01-04 Thread Magnus Löfqvist


4 jan 2013 kl. 23:59 skrev Danny Nicholas 
da...@debsinc.commailto:da...@debsinc.com:


From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus Löfqvist
Sent: Friday, January 04, 2013 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Calender and EWS with shared calenders

Hi all,

We want to use res_calender_ews to close users extensions if there are busy in 
there exchange calenders.

It is possible to use shared calenders ? Ie, I have a resource user that have 
access to the users calenders, so I dont need to maintain every users 
password/username in asterisk.

We are today running 1.8 but are going to upgrade to 11...

Best regards / Magnus

If it works for you in 1.8 it should work in 11.X – the only change is the 
$Revision number.


Dont know if it workes in 1.8. Havent set it up yet.
My question is if it works in 1.8 (or if it will work in 11)

/ Magnus
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Re: [asterisk-users] [Spam] Re: Problems with AGI and existing channel

2012-10-18 Thread Magnus Löfqvist
Hi,

The script is not essentials in this.
I have tried with a empty script... 
It stills generate problems with existing channels.

But if a try with a empty perl script, it dosent... seems to be something that 
php is doing when starting up.



Best regards
Magnus 


-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] För A J Stiles
Skickat: den 18 oktober 2012 13:57
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: [Spam] Re: [asterisk-users] Problems with AGI and existing channel

On Thursday 18 October 2012, Magnus Löfqvist wrote:
 We have some problem when running a AGI script (build with PHP), 
 existing channels (all of them) gets a hickup and then continues. We 
 are using AGI to lookup incoming calls in directory.
 
 It is kinda annoying, and I don't understand how it can be related.

I think you forgot to attach the actual script.  Without that, we can't help 
you.

--
AJS

Answers come *after* questions.

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[asterisk-users] Spam

2010-11-24 Thread Cary Fitch
I have been pounded with new, mostly text spam in the last few weeks.
Tonight I realized that the address that is being spammed is a personal one
I use for this list.

Has anyone else noticed new spam in the last 2-3 weeks?

Cary Fitch


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Re: [asterisk-users] Spam

2010-11-24 Thread Doug Lytle
Cary Fitch wrote:
 Has anyone else noticed new spam in the last 2-3 weeks?


No,

But I run ASSP in front of my MTA.

Doug

-- 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Spam

2010-11-24 Thread Steven Stromer
Same here. But, can the genie ever be put back in the bottle?

 Cary Fitch wrote:
 Has anyone else noticed new spam in the last 2-3 weeks?


 No,

 But I run ASSP in front of my MTA.

 Doug

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Re: [asterisk-users] spam blacklist

2010-07-29 Thread Benny Amorsen
SIP s...@arcdiv.com writes:

 Spammers sign up to the Asterisk mailing list and send spam once in a 
 while. My spam filter rejects it, and bounces the emails back to the 
 Asterisk list, which then drops me from the list because it got a single 
 bounce.

Don't ever bounce spam! You WILL get blacklisted for doing so. Either
reject it in the transaction or drop it on the floor.


/Benny


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Re: [asterisk-users] spam blacklist

2010-07-29 Thread SIP
  Supposedly, the filters drop it in the transaction stage. But for some 
reason, every time I get dropped from the list, it's just after a spam 
email was sent out en masse, so I'm not sure what's up there.

On 7/28/10 10:43 PM, jon pounder wrote:
 SIP wrote:
 what can you do ? simple discard spam don't bounce it.
On 7/28/10 9:45 PM, Sam wrote:

 Just a note, the asterisk mailing list server continually gets
 blacklisted over at
 http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
 mail to spamtraps. Perhaps something needs to be looked into...

 Regards,
 Sam


 Spammers sign up to the Asterisk mailing list and send spam once in a
 while. My spam filter rejects it, and bounces the emails back to the
 Asterisk list, which then drops me from the list because it got a single
 bounce.

 Bit of a pain in the left ventricle, really, but what can you do.


 N.





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[asterisk-users] spam blacklist

2010-07-28 Thread Sam
Just a note, the asterisk mailing list server continually gets 
blacklisted over at 
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering 
mail to spamtraps. Perhaps something needs to be looked into...

Regards,
Sam

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Re: [asterisk-users] spam blacklist

2010-07-28 Thread Kyle Kienapfel
My guess is on spammers signing up the spamtraps for mailing lists ;)

On Wed, Jul 28, 2010 at 6:45 PM, Sam aster...@net153.net wrote:
 Just a note, the asterisk mailing list server continually gets
 blacklisted over at
 http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
 mail to spamtraps. Perhaps something needs to be looked into...

 Regards,
 Sam

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Re: [asterisk-users] spam blacklist

2010-07-28 Thread jon pounder
SIP wrote:
what can you do ? simple discard spam don't bounce it.
   On 7/28/10 9:45 PM, Sam wrote:
   
 Just a note, the asterisk mailing list server continually gets
 blacklisted over at
 http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
 mail to spamtraps. Perhaps something needs to be looked into...

 Regards,
 Sam

 
 Spammers sign up to the Asterisk mailing list and send spam once in a 
 while. My spam filter rejects it, and bounces the emails back to the 
 Asterisk list, which then drops me from the list because it got a single 
 bounce.

 Bit of a pain in the left ventricle, really, but what can you do.


 N.

   


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Re: [asterisk-users] spam blacklist

2010-07-28 Thread SIP
  On 7/28/10 9:45 PM, Sam wrote:
 Just a note, the asterisk mailing list server continually gets
 blacklisted over at
 http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
 mail to spamtraps. Perhaps something needs to be looked into...

 Regards,
 Sam

Spammers sign up to the Asterisk mailing list and send spam once in a 
while. My spam filter rejects it, and bounces the emails back to the 
Asterisk list, which then drops me from the list because it got a single 
bounce.

Bit of a pain in the left ventricle, really, but what can you do.


N.

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[asterisk-users] Spam and that recent 'attack' ...

2010-04-16 Thread Gordon Henderson

Here's an intersting thing - in my spam folder today in popped an email 
from a Chinese manufacturing company trying to sell me the usual VoIP 
stuff - phones, etc. (From com-vox.com FWIW).

However as well as targetting the email address on my web site, they also 
targetted a made-up, but plausable, email address at the host that was 
attacked last weekend - (syst...@drogon.net and syst...@x.drogon.net) 
and that host has no public mentions anywhere that I'm aware of (Although 
once upon a time I did publish a SIP URI to it, but that was removed well 
over a year ago). Also that host doesn't run email and has no MX records 
pointing to it.

So it's probably not related, but just a passing curiosity...

Gordon

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Re: [asterisk-users] [SPAM] - Asterisk + Sip Phone + BLF - Email found in subject

2010-03-16 Thread DLeese
 ... and Snom (370) 
 SIP Phones, but with 2 extensions BLF status does not work correctly.
 have someone ever tested a Sip Phone with more then 60 BLF 
 without problems?
 Can someone suggest me model and brand?
 
 Thanks, bye. 

Here at our installation with over 100 snom320 we also encounter
problems with extensive usage of the BLF. In particular the receptionist
phone which uses multiple lines, appr. 30 BLF and hold+transfer.

Unfortunatly i can't suggest any phones, but htere are several things
you can do to optimiz the BLF:

http://wiki.snom.com/FAQ/When_using_%E2%80%9EExtension_Monitoring/Busy_L
amp_Field%E2%80%9C_on_more_than_1_Expansion_Module%2C_the_phone_response
_becomes_very_slow.

It is also possible to get a firmware which is supposed to fix the BLF
issues via the snom support (e.g.
snom320-7.3.latest_snapshot_snom-SIP-f.bin for 320's).

Perhaps this info can help you a a little.

Regards

Daniel

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Re: [asterisk-users] [SPAM] - Re: Asterisk t38modem Fax gateway evaluation - Email found in subject

2010-02-19 Thread DLeese
Hi,

Many thanks Steve and Philipp for your input. I am compiling Asterisk
1.6  svn version right now and will try to integrate the T.38 Gateway
patches mentioned at https://issues.asterisk.org/view.php?id=13405 (I
have some Linux coding experience, but unfortunately my
telecommunication knowledge is severly lacking).

The ISDN PRI - Analogue Converter solution mentioned by Philipp is
impractical for my situation as the hub of the 2-wire POTS lines is not
in the same Building as the PRI.

I will let you her if my research bears some practical results, as this
seems to be something not very well established/documented.

Daniel


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Re: [asterisk-users] [SPAM:9.0] extension not found

2010-02-13 Thread Brian
On Sat, 2010-02-13 at 12:54 +0530, cool dude wrote:
 hi friend need ur help in dial plan, i want to allow exten 2000 to
 2005 can make call outside and exten 2006 to 2010 can not make call
 outside. heres my dial plan.
  
 sip.conf
  
 [general]
 port = 5060
 bindaddr = 0.0.0.0
 context = others
 [2000]
 type=friend
 context=outside
 secret=1234
 host=dynamic
 
 [2001]
 type=friend
 context=outside
 secret=1234
 host=dynamic
 
 [2002]
 type=friend
 context=outside
 secret=1234
 host=dynamic
 
 [2003]
 type=friend
 contex=outside
 secret=1234
 host=dynamic
  
 [2004]
 type=friend
 contex=outside
 secret=1234
 host=dynamic
 
 [2005]
 type=friend
 contex=outside
 secret=1234
 host=dynamic
  
 [2006]
 type=friend
 contex=internal
 secret=1234
 host=dynamic
 
 [2007]
 type=friend
 contex=internal
 secret=1234
 host=dynamic
 [2008]
 type=friend
 contex=internal
 secret=1234
 host=dynamic
 
 [2009]
 type=friend
 contex=internal
 secret=1234
 host=dynamic
 
 [2010]
 type=friend
 contex=internal
 secret=1234
 host=dynamic
 
 ##
  ##
 vi /etc/asterisk/extensions.conf
 [from-zaptel]
 exten = s,1,wait(2)
 exten = s,n,dial(sip/2000)
 exten = s,n,dial(sip/2001)
 exten = s,n,Playback(tt-weasels)
 [others]
 include = internal
 include = outside
 [internal]
 exten = _20XX,1,Dial(SIP/${EXTEN})
 exten = _20XX,n,VoiceMail(${ext...@others,u)
 exten = _20XX,n,Hangup()
 [outside]
 exten = 2001,1,Dial(Zap/1-1/${EXTEN})
 exten = 2001,n,Hangup
 exten = 2002,1,Dial(Zap/1-1/${EXTEN})
 exten = 2002,n,Hangup
 exten = 2003,1,Dial(Zap/1-1/${EXTEN})
 exten = 2003,n,Hangup
 exten = 2004,1,Dial(Zap/1-1/${EXTEN})
 exten = 2004,n,Hangup
 exten = 2005,1,Dial(Zap/1-1/${EXTEN})
 exten = 2005,n,Hangup
 
 this is the log when i am calling from exten 2000 to outside
 
 Connected to Asterisk 1.4.29 currently running on localhost (pid =
 2243)
 Verbosity is at least 3
 [Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124
 handle_request_invite: Call from '2002' to extension '9193696136'
 rejected because extension not found.
 

You don't have an extension '9193696136' or anything to handle it. I'm
not familiar with your dialcodes, so assuming you are dialing '9' for an
outside line, followed by a 9 digit number this may work (untested)


exten = _9XX,1,NoOp(--NOTED OUTSIDE CALL--)
exten = _9XX,2,Dial(ZAP/1-1/${EXTEN:1})


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Re: [asterisk-users] [SPAM:9.0] extension not found

2010-02-13 Thread Tzafrir Cohen
On Sat, Feb 13, 2010 at 07:58:39AM +, Brian wrote:

 
 You don't have an extension '9193696136' or anything to handle it. I'm
 not familiar with your dialcodes, so assuming you are dialing '9' for an
 outside line, followed by a 9 digit number this may work (untested)
 
 
 exten = _9XX,1,NoOp(--NOTED OUTSIDE CALL--)
 exten = _9XX,2,Dial(ZAP/1-1/${EXTEN:1})

Also: 

ZPA/1 rather than ZAP/1-1

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] [SPAM:9.0] extension not found

2010-02-13 Thread Doug Lytle
Tzafrir Cohen wrote:
 On Sat, Feb 13, 2010 at 07:58:39AM +, Brian wrote:

 ZPA/1 rather than ZAP/1-1



That'd be ZAP/1

Doug



-- 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] [SPAM:9.0] extension not found

2010-02-13 Thread Brian
On Sat, 2010-02-13 at 07:03 -0500, Doug Lytle wrote:
 Tzafrir Cohen wrote:
  On Sat, Feb 13, 2010 at 07:58:39AM +, Brian wrote:
 
  ZPA/1 rather than ZAP/1-1
 
 
 
 That'd be ZAP/1
 
 Doug
 
Thank you Doug - that brightened up my day * 10 - being dyslexic I'd not
spotted that in the first instance.


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[asterisk-users] {Spam?} MeetMe/Dahdi for FreeBSD

2010-01-05 Thread Leif Neland
It seems dahdi is needed for meetme, but not available under FreeBSD.
So what do I do then?
Asterisk has only SIP-channels.



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Re: [asterisk-users] *****SPAM***** sched_settime: Request to schedule in the past?!?!

2009-10-19 Thread Danny Nicholas
Logically this must mean that the timestamp on the request is before the
current system time.

This could be caused by an out-of-sync condition between the system and
hardware clocks

 

http://forum.voxilla.com/asterisk-support-forum/asterisk-warning-reschedule-
past-9177.html

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Fisher
Sent: Saturday, October 17, 2009 10:18 AM
To: asterisk-users@lists.digium.com
Subject: *SPAM* [asterisk-users] sched_settime: Request to schedule
in the past?!?!

 

Does anybody know what this message means?

 

[2009-10-17 08:11:59] DEBUG[9598]: sched.c:204 sched_settime: Request to
schedule in the past?!?!

 

I've tried all I can find on google with no change

 

It seems to happen when call is at a READ()  or Background() - What the
caller hear is small delays after a press during a message.

 

Still using asterisk 1.4.22.1 because it seems to have the best DTMF
detection during a playback

 

Thanks, Bart

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Re: [asterisk-users] *****SPAM***** DTMF problem during read()

2009-10-05 Thread Danny Nicholas
You are playing the prompt with Background or Playback?  Please post the
dialplan snippet.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Fisher
Sent: Monday, October 05, 2009 10:03 AM
To: asterisk-users@lists.digium.com
Subject: *SPAM* [asterisk-users] DTMF problem during read()

 

I have a simple dialplan.  Using the read cmd, I ask caller for his
passcode.  If the caller is calling from an plain old analog phone, his dtmf
is not heard until the prompt stops playing. SIP phones work correctly. I've
trird everything I found searching the net. I've tried all the dtmfmode. I'm
using 1.4.26

 

Currently my vitelity sip account is setup:

 

insecure=very
canreinvite=no
host=xx.xx.xx.xx
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
rfc2833compensate=yes

I need to trouble shoot this furthur.  I read I can enable rtp debug IP
but I can't find any output.

 

Thanks, Bart

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Re: [asterisk-users] *****SPAM***** Method to downgrade asterisk

2009-10-05 Thread Danny Nicholas
Each is independent of the other.  The important things are to make sure
asterisk is not running when doing make install and to clean
/usr/lib/asterisk/modules before make install.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Fisher
Sent: Monday, October 05, 2009 11:54 AM
To: asterisk-users@lists.digium.com
Subject: *SPAM* [asterisk-users] Method to downgrade asterisk

 

I currently have asterisk-1.4.26.2 installed and working.  It was sugguested
I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems.

 

What is the method to downgrade?

 

Do I just do in the asterisk-1.4.25 folder:

 

make clean
./configure
make install

 

Or do I need to 'make clean' in the asterisk-1.4.26.2 first then move to the
asterisk-1.4.25 folder and do ./configure  make install?

 

Thanks, Bart

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

2009-09-20 Thread Kev Szaszvari
Or you could not cheat and do it properly

- Original Message -
From: C. Savinovich
[mailto:c.savinov...@itntelecom.com]
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' [mailto:asterisk-us...@lists.digium.com]
Sent:
Thu, 17 Sep 2009 19:30:26 +1000
Subject: [asterisk-users] [SPAM] RE:  dCAP
Exam


 What about if I use the browser from my cellular phone?
 
  
 
 CS
 
  
 
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
 Sent: Wednesday, September 16, 2009 10:21 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] dCAP Exam
 
  
 
 I believe the administrator can see what is on your screen with screen with
 those screen sharing stuff, this makes it harder a lil bit, and
 www.boratproxy.com becomes useless in that case.
 
  
 
 On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro
 stot...@totarotechnologies.com wrote:
 
  
 
 On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher tles...@digium.com
 wrote:
 
 On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
  Hmm...so by open book, that means access to the internet? Possible to
  get own notes ?
 
 Yes, you have access to the Internet, but your access is proxied, and the
 administrator of the test can see everything that you access.  So it's best
 for you stick with only general guides and not look for crib notes.  If your
 test proctor believes you cheated, you fail.
 
 
 --
 Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com  www.asterisk.org
 
 
 Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
 dump sites.   
 
 Or go to www.boratproxy.com and confuse their proxy.  ah too fun.
 
 
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 -- 
 Pascal B.
 http://www.kameleonlabs.com/
 Mike Ditka http://www.brainyquote.com/quotes/authors/m/mike_ditka.html   -
 If God had wanted man to play soccer, he wouldn't have given us arms. 
 
 

This Communication is intended only for the use of the individual or entity to 
which it is addressed and may contain information that is privileged, 
confidential or copyright. You are hereby notified that any dissemination, 
distribution or copying of this communication is
strictly prohibited without the authority of the sender. If you have received 
this e-mail message in error or are not the intended recipient, please delete 
and destroy all copies and notify us immediately by return mail. Any views 
expressed in this communication are those 
of the individual sender, except where the sender specifically states 
otherwise. If you no longer want to receive notifications, simply reply to this 
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[asterisk-users] [SPAM] RE: dCAP Exam

2009-09-16 Thread C. Savinovich
What about if I use the browser from my cellular phone?

 

CS

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, September 16, 2009 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dCAP Exam

 

I believe the administrator can see what is on your screen with screen with
those screen sharing stuff, this makes it harder a lil bit, and
www.boratproxy.com becomes useless in that case.

 

On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro
stot...@totarotechnologies.com wrote:

 

On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher tles...@digium.com
wrote:

On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
 Hmm...so by open book, that means access to the internet? Possible to
 get own notes ?

Yes, you have access to the Internet, but your access is proxied, and the
administrator of the test can see everything that you access.  So it's best
for you stick with only general guides and not look for crib notes.  If your
test proctor believes you cheated, you fail.


--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org


Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
dump sites.   

Or go to www.boratproxy.com and confuse their proxy.  ah too fun.


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-- 
Pascal B.
http://www.kameleonlabs.com/
Mike Ditka http://www.brainyquote.com/quotes/authors/m/mike_ditka.html   -
If God had wanted man to play soccer, he wouldn't have given us arms. 

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

2009-09-16 Thread Jared Smith
On Thu, 2009-09-17 at 14:00 +0430, C. Savinovich wrote:
 What about if I use the browser from my cellular phone?

Sorry, cell phone use is not permitted during the testing.  We've had
students try to snap pictures of the exam with their cell phone cameras,
so we had to institute a policy against cell phone use.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] [SPAM] RE: dCAP Exam

2009-09-16 Thread John Millican
C. Savinovich wrote:
 What about if I use the browser from my cellular phone?
 
  
 
 CS
 
  
 
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
 Sent: Wednesday, September 16, 2009 10:21 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] dCAP Exam
 
  
 
 I believe the administrator can see what is on your screen with screen with
 those screen sharing stuff, this makes it harder a lil bit, and
 www.boratproxy.com becomes useless in that case.
 
  
 
 On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro
 stot...@totarotechnologies.com wrote:
 
  
 
 On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher tles...@digium.com
 wrote:
 
 On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
 Hmm...so by open book, that means access to the internet? Possible to
 get own notes ?
 
 Yes, you have access to the Internet, but your access is proxied, and the
 administrator of the test can see everything that you access.  So it's best
 for you stick with only general guides and not look for crib notes.  If your
 test proctor believes you cheated, you fail.
 
 
 --
 Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com  www.asterisk.org
 
 
 Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain
 dump sites.   
 
 Or go to www.boratproxy.com and confuse their proxy.  ah too fun.
 
I can't resist:
After having taken many MS and Cisco tests in the past, it would seem
rather apparent to me that for the dcap, as with any other test, if you
know what you are doing you are all set and you don't have to try and
find ways to cheat that don't  look like cheating.

Disclaimer: I have not taken the dcap test yet!

JohnM


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Re: [asterisk-users] [SPAM] RE: dCAP Exam

2009-09-16 Thread Zoaaaaa

 What if i send my twin brother to take the exam instead of me... ?

z

C. Savinovich wrote:

 What about if I use the browser from my cellular phone?

  

 CS

  

 *From:* asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal 
 Bruno
 *Sent:* Wednesday, September 16, 2009 10:21 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] dCAP Exam

  

 I believe the administrator can see what is on your screen with screen 
 with those screen sharing stuff, this makes it harder a lil bit, and 
 www.boratproxy.com http://www.boratproxy.com becomes useless in that 
 case.

  

 On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro 
 stot...@totarotechnologies.com 
 mailto:stot...@totarotechnologies.com wrote:

  

 On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher tles...@digium.com 
 mailto:tles...@digium.com wrote:

 On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote:
  Hmm...so by open book, that means access to the internet? Possible to
  get own notes ?

 Yes, you have access to the Internet, but your access is proxied, and the
 administrator of the test can see everything that you access.  So it's 
 best
 for you stick with only general guides and not look for crib notes. 
  If your
 test proctor believes you cheated, you fail.


 --
 Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com http://www.digium.com  
 www.asterisk.org http://www.asterisk.org


 Just tunnel your HTTP traffic over an SSH link and go to some dCAP 
 brain dump sites.  

 Or go to www.boratproxy.com http://www.boratproxy.com and confuse 
 their proxy.  ah too fun.


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 -- 
 Pascal B.
 http://www.kameleonlabs.com/
 Mike Ditka 
 http://www.brainyquote.com/quotes/authors/m/mike_ditka.html  - If 
 God had wanted man to play soccer, he wouldn't have given us arms.

 

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Re: [asterisk-users] [SPAM] RE: dCAP Exam

2009-09-16 Thread Pascal Bruno
On Wed, Sep 16, 2009 at 2:37 PM, Zoa zoach...@securax.org wrote:


  What if i send my twin brother to take the exam instead of me... ?

 z


If you think you cannot pass the test yourself, your twin wont be able to
pass it neither, he can be even worst than you

lol
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[asterisk-users] SPAM

2009-07-17 Thread Doug Lytle
I seem I'm getting pelted with the UK Pharmacy Online Sale 80 SPAM 
again, I'm looking forward to being kicked off the list again shortly.

*sigh*

Doug

-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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[asterisk-users] [SPAM] RE: newbie questions

2009-06-20 Thread C. Savinovich
Let me see if I get you: you inserted the installation CD, then you
restarted the computer, and now you want to know what to do next?

CS

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Poe
Sent: Friday, June 19, 2009 11:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] newbie questions

I have an Asterisknow.org CD.  When I boot up, it seems ready for me to 
choose update, console, etc.  I'm assuming I need to do something at the 
CLI prompt.  Is there a tutorial that would take me from loading CD to 
making first test call?

Computer is Dell Optiplex GX260
50GB free disk space
1.5GB RAM
P4 processor
external mic
speakers
Skype is on board, and would be good to use it, if possible. 

If I want to use Skype, do I need anything additional?  Would it be 
better to install CD on my hard drive?  Any help appreciated.
Tom

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Re: [asterisk-users] [SPAM] RE: newbie questions

2009-06-20 Thread Tom Poe
C. Savinovich wrote:
 Let me see if I get you: you inserted the installation CD, then you
 restarted the computer, and now you want to know what to do next?

 CS

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Poe
 Sent: Friday, June 19, 2009 11:25 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] newbie questions

 I have an Asterisknow.org CD.  When I boot up, it seems ready for me to 
 choose update, console, etc.  I'm assuming I need to do something at the 
 CLI prompt.  Is there a tutorial that would take me from loading CD to 
 making first test call?

 Computer is Dell Optiplex GX260
 50GB free disk space
 1.5GB RAM
 P4 processor
 external mic
 speakers
 Skype is on board, and would be good to use it, if possible. 

 If I want to use Skype, do I need anything additional?  Would it be 
 better to install CD on my hard drive?  Any help appreciated.
 Tom

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Hello:  Cancel these questions.  It's looking more and more like I need 
to have a dedicated computer for proper Asterisk implementation, even 
for a home single line user.  I found the link for ordering O'Reilly 
Asterisk book.  Will spend time with that, first.  Thanks for those who 
have responded.  By the way, my AsteriskNow CD had no install guide with it.
Tom

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[asterisk-users] [SPAM] RE: SIP hacked connection?

2009-06-11 Thread C. Savinovich
Very few calls have been made this way, trivial cost, but it is slightly
worrying.

  That's what I thought when they hacked into one of my systems, but it is
not the cost of the calls, it is the purposed of the calls you should watch
out for.  The FBI contacted the owner of the PBX, and inquired him about
calls being made from his company doing credit card soliciting.

CS


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Redstone
Sent: Thursday, June 11, 2009 3:30 PM
To: Asterisk User
Subject: [asterisk-users] SIP hacked connection?

Hi

Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface
(Junghanns).

SIP ports mapped through firewall as we often connect from outside, but all
SIP accounts have good passwords.

However our telecoms provider picked up a few suspicious calls to places we
do not normally call at times we do not often call.

Looking at Asterisk logs it shows SIP session from the internet connected in
and making calls with account IDs we do not recognise - definitely none of
ours.

Very few calls have been made this way, trivial cost, but it is slightly
worrying.

Anyone any ideas on how this could be happening?

Thank

Paul


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Re: [asterisk-users] {Spam?} Vacation reply

2009-04-14 Thread Hans Witvliet
On Tue, 2009-04-14 at 09:45 -0700, awerf...@hotmail.com wrote:
 Hi Friend,
 
 How are you doing recently?

I'm getting bored.

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Re: [asterisk-users] ***SPAM*** Re: Asterisk-beginner : cannot make phonecalls using Asterisk

2009-04-13 Thread ROQUÉ, Francisco Emiliano
In your sip.conf or sip_nat.conf (for elastix) set the variables:

externhost=
externip=
domain=
externrefresh=
localnet=

Regards

Francisco



Anthony Plack wrote:
 bindaddr = 0.0.0.0

 

 I would set this to the ethernet interface IP address, I believe this may be 
 your issue.

 Registration is only for receiving calls, if you are not seeing information 
 on the dial, then the phone is not talking to the server.  I would make sure 
 of the settings in the web-interface as well.

 Tony Plack



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[asterisk-users] [SPAM] RE: $20 Bounty

2009-03-03 Thread C. Savinovich
 

  Dear Sir:

 

 Me no peak englisss…. 20 pesos???  ok , tank you sir, tank you sir

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kinjal Dixit
Sent: Tuesday, March 03, 2009 9:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] $20 Bounty

 

 


I wish my family and I could live on $40 a week...

 


simplify, simplify, simplify

-- 
http://www.linkedin.com/in/kinjaldixit

open networker

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Re: [asterisk-users] [SPAM] enabling silence suppression in asterisk

2009-01-06 Thread Lyle Giese
bala krishnan wrote:

 Hi Friends,
 Currently i am using the asterisk 1.4.x version. In that i want to
 enable to silence suppression in the SIP calls. Please tell me the
 configuration changes to be done.



 Thanks in advance,
 balasam.



Enabling silence suppression is a bad thing.  Asterisk and sip phone
will think that the other party has dropped off and will randomly drop
calls on you because of a lack of traffic from the other party.

Lyle


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Re: [asterisk-users] [SPAM] - MySQL Error Message - Email found in subject

2008-12-02 Thread Andrew Thomas
Give this a go:

 

exten = s,n,MYSQL(Query resultid ${connid} SELECT `name` FROM `cnam`
WHERE `ani` = '${CALLERID(number)}')

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Re: [asterisk-users] [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject

2008-12-01 Thread Andrew Thomas
It looks like you are trying to dial out on your 'NT' instead of your
'TE'.

 

Try changing Dial(DAHDI/g1/${EXTEN:1}); to Dial(DAHDI/G1/${EXTEN:1});

 

Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.

 

HTH

 

 

 

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Re: [asterisk-users] [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject

2008-12-01 Thread Olivier
2008/12/1 Andrew Thomas [EMAIL PROTECTED]

  It looks like you are trying to dial out on your 'NT' instead of your
 'TE'.



 Try changing Dial(DAHDI/*g*1/${EXTEN:1}); to Dial(DAHDI/*G*1/${EXTEN:1});

Unfortunately, changing g1 to G1 didn't change end result.
Anyway, I can't really see why using NT or TE port should theorically change
end result as both allow incoming and outgoing calls.
Do you agree ?



 Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.



 HTH







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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject - Email found in subject

2008-12-01 Thread Andrew Thomas
Apart from you were dialling out on your inbound context and
vice-versa.

 

The best advice I can give now is to change to mISDN - as this
is proven to work with v1.4 and v1.6.

 

Actually - have you tried putting the 100ohm termination on for
your NT port?

 

I need to do that with mISDN as it only allows ptmp for ISDN
extensions.

 

Cheers

Andy

 

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Re: [asterisk-users] [SPAM] - Re: CDR Design - Email found in subject

2008-12-01 Thread Andrew Thomas
Hi murf,

Speaking as someone who designs and builds billing platforms, this is
very exciting.

One little thing I have most problems with is the good old fax
detection.  I know that NVFaxDetect et al do actually answer the call
and, therefore, get flagged as ANSWERED in the CDR.

But, if the call never gets answered after the initial detection - then,
to my customer, it is a missed (NO ANSWER) call.

Is there anyway of following that scenario?

In other words:

 Call comes in over DAHDI, ZAP or mISDN channel
 NVFaxDetect samples call
 It's not fax, so call is passed on...
 Call never gets answered and caller hangs up.

I appreciate that the NV suite are not Digium products - but it would be
nice if the new CDR information could at least flag it up as a 'type',
so that I can manually (well, via. my code) trace the call through to
check for human intervention.

Actually, the same would go for v1.6 and it's built in fax detection :).

I hope that makes sense.

Cheers
Andy


Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF 


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Re: [asterisk-users] [SPAM] - Re: CDR Desgin - Email found in subject

2008-12-01 Thread Andrew Thomas
...or something along the lines of a setting a variable (like we do for
MONITOR_EXEC)...



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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject - Email found in subject

2008-12-01 Thread Olivier
2008/12/1 Andrew Thomas [EMAIL PROTECTED]

   Apart from you were dialling out on your inbound context and vice-versa.



 The best advice I can give now is to change to mISDN – as this is proven to
 work with v1.4 and v1.6.

 I wanted to try the Digium B410P card with Dahdi as those are now supported
in 1.6.X.
Hopefully, it would simplify software configuration as both asterisk and
dahdi are supported by Asterisk community.
As soon as I can get this card running with dahdi, I will give a try with
mISDN.

Fortunately, you still can use misdn with B410P w/ 1.6.X.



 Actually – have you tried putting the 100ohm termination on for your NT
 port?

 I didn't dare to try as I'm not familiar with concepts behind that.



 I need to do that with mISDN as it only allows ptmp for ISDN extensions.



 Cheers

 Andy



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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject

2008-12-01 Thread Andrew Thomas
The DAHDI docs actually include the 'steps' that used to be in Zaptel's
'make b410p'.  These steps involve downloading and compiling mISDN.  Why
re-invent the wheel?

 

Just a thought :-).

 

As for the 100ohm termination bit - it's simply changing a couple of dip
switches on the b410p card (as described in the manual).

 

Andy

 

 

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: 01 December 2008 17:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM] - Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi,b410p
and looping from 1 port to another - Email found in subject -Email found
in subject - Email found in subject

 

 

2008/12/1 Andrew Thomas [EMAIL PROTECTED]

Apart from you were dialling out on your inbound context and
vice-versa.

 

The best advice I can give now is to change to mISDN - as this
is proven to work with v1.4 and v1.6.

I wanted to try the Digium B410P card with Dahdi as those are now
supported in 1.6.X.
Hopefully, it would simplify software configuration as both asterisk and
dahdi are supported by Asterisk community.
As soon as I can get this card running with dahdi, I will give a try
with mISDN.

Fortunately, you still can use misdn with B410P w/ 1.6.X.

 

Actually - have you tried putting the 100ohm termination
on for your NT port?

I didn't dare to try as I'm not familiar with concepts behind that.

 

I need to do that with mISDN as it only allows ptmp for
ISDN extensions.

 

Cheers

Andy

 


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Re: [asterisk-users] [SPAM] - Re: [SPAM] - Re: [SPAM] - Dahdi, b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject

2008-12-01 Thread Olivier
2008/12/1 Andrew Thomas [EMAIL PROTECTED]

  The DAHDI docs actually include the 'steps' that used to be in Zaptel's
 'make b410p'.  These steps involve downloading and compiling mISDN.  Why
 re-invent the wheel?

in 1.6.1rc5, mISDN is optional ...



 Just a thought J.

I will compare both ASAP ...



 As for the 100ohm termination bit – it's simply changing a couple of dip
 switches on the b410p card (as described in the manual).

That's the point : why is it necessary to change to 100ohm termination ?



 Andy







 -Original Message-
 *From:* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *On Behalf Of *Olivier
 *Sent:* 01 December 2008 17:54
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [SPAM] - Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi,b410p
 and looping from 1 port to another - Email found in subject -Email found in
 subject - Email found in subject





 2008/12/1 Andrew Thomas [EMAIL PROTECTED]

 Apart from you were dialling out on your inbound context and vice-versa.



 The best advice I can give now is to change to mISDN – as this is proven to
 work with v1.4 and v1.6.

   I wanted to try the Digium B410P card with Dahdi as those are now
 supported in 1.6.X.
 Hopefully, it would simplify software configuration as both asterisk and
 dahdi are supported by Asterisk community.
 As soon as I can get this card running with dahdi, I will give a try with
 mISDN.

 Fortunately, you still can use misdn with B410P w/ 1.6.X.



 Actually – have you tried putting the 100ohm termination on for your NT
 port?

I didn't dare to try as I'm not familiar with concepts behind that.



 I need to do that with mISDN as it only allows ptmp for ISDN extensions.



 Cheers

 Andy




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Re: [asterisk-users] [SPAM] - Re: FW: cdr_addon_mysql.so did notregister itselfduringload - Email found in subject

2008-11-28 Thread Andrew Thomas
Did you install the MySQL libraries?

Debian's command is - apt-get install libmysqlclient15-dev

Andy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthias
Urlichs
Sent: 27 November 2008 16:05
To: asterisk-users@lists.digium.com
Subject: [SPAM] - Re: [asterisk-users] FW: cdr_addon_mysql.so did
notregister itselfduringload - Email found in subject

On Thu, 28 Dec 2006 12:34:46 -0600, Savoy, Kevin - Williston, ND wrote:

 checking for mysql_init in -lmysqlclient... no
 
 What do I need to make that say yes?

You need to read config.log and check _why_ the link fails.


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Re: [asterisk-users] [SPAM] - Asterisk and S-Bus - Email found in subject

2008-11-28 Thread Andrew Thomas
Have you set port 2 as 'NT' in the mISDN config file (not the Asterisk one)?

Also, you will probably need to set it to ptmp.

You need to configure them in misdn.conf (the Asterisk one this time).

Here's the tail of my misdn.conf (4 x BRI):

[trunks]
ports = 1,2 ; physical port numbers (as defined in mISDN.conf)
context = inbound ; context for incoming calls in extensions.conf
msns = * ; accept and process every number that comes in and let my 
extensions.conf sort it out (easiest way)

[extensions]
ports = 3,4 ; physical port numbers (as defined in mISDN.conf)
context = default ; context for internal users in extensions.conf

Ports 1 and 2 are my trunk lines - ports 3 and 4 are my ISDN modems.

Hope that helps.

Andy

 -Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: 28 November 2008 11:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM] - [asterisk-users] Asterisk and S-Bus - Email found in subject


Hi everyone, 

I've built an Asterisk server (1.4.22) on a Debian Etch 4.0 base system (Kernel 
2.6.18). I have so far installed AsteriskGUI and the Zaptel, libPRI and mISDN 
drivers. 

The hardware is a dual processor, dual core Xeon 2ghz (per core) server with 
one Digium Wildcard B410P (4 FXS - 4 FXO) and one Beronet BN2S0 ISDN card (1 TE 
- 1 NT). 

The all is working well, however I am unsure of how to configure the BN2S0 card 
port 2 (NT) to provide an S-Bus that additional ISDN phones can connect to as 
extensions. The phone is powering up OK and Asterisk is dectecting when the 
phone is taken off hook however I receive an error about Asterisk not knowing 
how to handle the call. 

As of yet, I have not configured any additional users in users.conf for the 
ISDN phone as Im not quite sure how to configure them to use that channel. 

Has anyone ever tried such a configuration here that might be able to give me a 
few pointers? Any and all help is much appreciated. 

Thanks. 

Kind Regards,
Steven Moughan
-
LAKE Communications,
Beech House, Greenhills Road,
Dublin 24, IRELAND
int. +353 1 4031112
fax. +353 1 452 0826
www.lakecommunications.com 

This email and any files transmitted with it are confidential and intended 
solely for the use of the individual or entity to whom they are addressed. If 
you have received this email in error please notify [EMAIL PROTECTED]
 
This footnote also confirms that this email message has been scanned for the 
presence of computer viruses and other security threats.
 
Registered Office: Lake Communications Ltd, Beech House, Greenhills Road, 
Dublin 24, Ireland.
Registered No. 59890

 

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[asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Gordon Henderson

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.

Gordon

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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread bails
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
 
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-- 
This message has been scanned for viruses and dangerous content
by MailScanner at Circlemail and is believed to be clean.


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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Steve Edwards
On Thu, 6 Nov 2008, 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.

The English have such a way with words :)

I keep a local archive of the last 30 days list posts. Searching for 
didforsale.com shows:

 Buy unmetered VoIP DID from  DidForSale.com

is the signature for Jai Rangi [EMAIL PROTECTED].

A wolf in the sheep's pen?

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread David Gibbons
I'm glad I'm not the only one who got that. I sent them a nasty response 
earlier this morning...



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Thursday, November 06, 2008 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

On Thu, 6 Nov 2008, 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.

The English have such a way with words :)

I keep a local archive of the last 30 days list posts. Searching for
didforsale.com shows:

 Buy unmetered VoIP DID from  DidForSale.com

is the signature for Jai Rangi [EMAIL PROTECTED].

A wolf in the sheep's pen?

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Gonzalo Servat
On Thu, Nov 6, 2008 at 2:11 PM, David Gibbons [EMAIL PROTECTED]wrote:

 I'm glad I'm not the only one who got that. I sent them a nasty response
 earlier this morning...


I got the same crap from them. I can't imagine anyone buying from a company
that spams subscribers of a mailing list to get business. But I guess spam
exists for a reason; there's always some moron out there that purchases
goods from these delinquents.

- Gonzalo
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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Anthony Francis
Gotta love this list being farmed for spammers now. I am sure they call 
it targeted delivery or some such nonsense. I can't wait for capitalism 
to completely fail, then there won't be any spam.

David Gibbons wrote:
 I'm glad I'm not the only one who got that. I sent them a nasty response 
 earlier this morning...



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
 Sent: Thursday, November 06, 2008 11:05 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

 On Thu, 6 Nov 2008, 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.
 

 The English have such a way with words :)

 I keep a local archive of the last 30 days list posts. Searching for
 didforsale.com shows:

  Buy unmetered VoIP DID from  DidForSale.com

 is the signature for Jai Rangi [EMAIL PROTECTED].

 A wolf in the sheep's pen?

 Thanks in advance,
 
 Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000

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-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP



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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Greg Woods
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
 Gotta love this list being farmed for spammers now. I am sure they call 
 it targeted delivery or some such nonsense. I can't wait for capitalism 
 to completely fail, then there won't be any spam.

Socialism has already completely failed. What should we do, go back to a
barter economy? :-)



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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Anthony Francis
http://en.wikipedia.org/wiki/Jacque_Fresco

A resource based economy.

Greg Woods wrote:
 On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
   
 Gotta love this list being farmed for spammers now. I am sure they call 
 it targeted delivery or some such nonsense. I can't wait for capitalism 
 to completely fail, then there won't be any spam.
 

 Socialism has already completely failed. What should we do, go back to a
 barter economy? :-)



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-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread David Gibbons
I think I'll take the occasional spam and keep my freedoms and civil 
liberties...

Tell Kim Jong Il I said hello though!

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Thursday, November 06, 2008 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
to completely fail, then there won't be any spam.

David Gibbons wrote:
 I'm glad I'm not the only one who got that. I sent them a nasty response 
 earlier this morning...



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
 Sent: Thursday, November 06, 2008 11:05 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

 On Thu, 6 Nov 2008, 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.


 The English have such a way with words :)

 I keep a local archive of the last 30 days list posts. Searching for
 didforsale.com shows:

  Buy unmetered VoIP DID from  DidForSale.com

 is the signature for Jai Rangi [EMAIL PROTECTED].

 A wolf in the sheep's pen?

 Thanks in advance,
 
 Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000

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--
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP



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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Singer X.J. Wang
He's dead, if you look at the recent photos of him his shadow is not 
where it should be compared to other people in the photos. Its all 
photoshop'ed now :)


David Gibbons wrote:

I think I'll take the occasional spam and keep my freedoms and civil 
liberties...

Tell Kim Jong Il I said hello though!

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Thursday, November 06, 2008 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
to completely fail, then there won't be any spam.

David Gibbons wrote:
  

I'm glad I'm not the only one who got that. I sent them a nasty response 
earlier this morning...



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Thursday, November 06, 2008 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

On Thu, 6 Nov 2008, 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.

  

The English have such a way with words :)

I keep a local archive of the last 30 days list posts. Searching for
didforsale.com shows:

 Buy unmetered VoIP DID from  DidForSale.com

is the signature for Jai Rangi [EMAIL PROTECTED].

A wolf in the sheep's pen?

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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--
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP



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--
*Singer Wang*
/System and Database Engineer/
The Pythian Group

Office: (613) 565-8696 x298
Toll Free:  (877) 798-4426 x298
Fax:(613) 565-8710
Email:  [EMAIL PROTECTED]
MSN:[EMAIL PROTECTED]
Yahoo:  pythianwang
AIM:pythianwang
ICQ:201253
Gadu-Gadu:  6817795
Tencent QQ: 858310404

begin:vcard
fn:Singer Wang
n:Wang;Singer
email;internet:[EMAIL PROTECTED]
tel;work:(613) 565-8696 x298
x-mozilla-html:TRUE
version:2.1
end:vcard

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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread SIP
Greg Woods wrote:
 On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
   
 Gotta love this list being farmed for spammers now. I am sure they call 
 it targeted delivery or some such nonsense. I can't wait for capitalism 
 to completely fail, then there won't be any spam.
 

 Socialism has already completely failed. What should we do, go back to a
 barter economy? :-)



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WHAT?? The telco has raised the cost of my call to Italy from 1 hen per
minute to 3! I'll be out of chickens in WEEKS if this keeps up. Next
thing you know, the cable co will start demanding 1.5 cows per month. Of
course, they don't care that I can't give them .5 cows without wasting a
WHOLE one. It's terrible! I only wish there were something we could use
for payment instead of commodities -- maybe some sort of note that took
the place of a physical commodity!

Hey...that sounds like a terrific idea!


N.

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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Brent Davidson
Singer X.J. Wang wrote:
 He's dead, if you look at the recent photos of him his shadow is not 
 where it should be compared to other people in the photos.

Well that's just lovely.  Kim Jong Il is now an immortal vampire.  
Better call the white house and tell them to replace the nuclear warhead 
with a bunch of garlic cloves and a wooden-stake claymore mine

.

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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]

2008-11-06 Thread Anthony Plack
LOL, I love people like Jacque who have no clue that wolves exist. The assumption that evil does not exist in the hearts of men is niavity.

What this discussion has to do with asteriskI have no clue... but entertaining none-the-less.
 http://en.wikipedia.org/wiki/Jacque_Fresco

 A resource based economy.

 Greg Woods wrote:
 On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:

 Gotta love this list being farmed for spammers now. I am sure
 they call it targeted delivery or some such nonsense. I can't
 wait for capitalism to completely fail, then there won't be any
 spam.


 Socialism has already completely failed. What should we do, go
 back to a barter economy? :-)


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 http://lists.digium.com/mailman/listinfo/asterisk-users


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[asterisk-users] Spam Filter

2008-06-30 Thread Andrew Joakimsen
Does anyone know of a spam filter that will work with Asterisk?

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Re: [asterisk-users] Spam Filter

2008-06-30 Thread Brian J. Murrell
On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote:
 Does anyone know of a spam filter that will work with Asterisk?

What does spam have to do with Asterisk?  Or do you mean spit perhaps?

http://en.wikipedia.org/wiki/VoIP_spam ?  Probably the same techniques
such as whilelisting, blacklisting and greylisting are going to have to
be applied.  It will be much more difficult however as there is no
digital form of SPIT that can be analysed before delivery and reported
to clearinghouses.

Then again, isn't SPIT just telemarketing and regulated by the same
(albeit jurisdictionally local) rules such as Do Not Call lists and so
on?

b.



signature.asc
Description: This is a digitally signed message part
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Re: [asterisk-users] Spam Filter

2008-06-30 Thread Andrew Joakimsen
Doesn't have to be Voip-originated.. One guy out of his apartment
in Texas constantly orders phone lines, disconnects them after a few
days of continuous dialing. http://customcampaigns.net/political.html
He is not calling with a political message but instead simply
harvesting phone numbers. What purpose Mr. Maxi does this for is
irrelevant, it's still illegal.

Point is SpamAssassin analyzes certain values in an email before its
delivered to determine if its spam or not. I am looking for something
that works in the same fashion but for phonecalls be they transmitted
via the PSTN, PRI, ISDN, SIP or IAX (etc)

If the telemarketers followed the laws this would not even be a concern.


On Mon, Jun 30, 2008 at 12:16 PM, Brian J. Murrell
[EMAIL PROTECTED] wrote:
 On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote:
 Does anyone know of a spam filter that will work with Asterisk?

 What does spam have to do with Asterisk?  Or do you mean spit perhaps?

 http://en.wikipedia.org/wiki/VoIP_spam ?  Probably the same techniques
 such as whilelisting, blacklisting and greylisting are going to have to
 be applied.  It will be much more difficult however as there is no
 digital form of SPIT that can be analysed before delivery and reported
 to clearinghouses.

 Then again, isn't SPIT just telemarketing and regulated by the same
 (albeit jurisdictionally local) rules such as Do Not Call lists and so
 on?

 b.


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Re: [asterisk-users] Spam: Voicemail- Recorded Mesage Low Volume

2008-04-01 Thread Brig C. McCoy
Do you see the same volume issues dialing in with a 'normal' POTS phone?

 

Are these the standard recordings that come with Asterisk or some custom
recordings?

 

...brig

 

Brig C. McCoy

ThyssenKrupp Access Corp

Network Administrator

Grandview, MO  64030

816-767-5577

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Meksavan
Sent: Tuesday, April 01, 2008 2:32 PM
To: Asterisk Users
Subject: Spam:[asterisk-users] Voicemail- Recorded Mesage Low Volume

 

Asterisk Users,

  I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a
Debian Etch system.  On the recorded voice mail messages, the volume
is really low when retrieving them with my cell phone.  I tried with
multiple cell phones with the volume level high and still, the same
problem. I tried to increase the rxgain to 12.2 in the zapata.conf file
and it had no affect on the volume.  So, tried to add the g option on
the asterisk voicemail command, like this:

exten = s-NOANSWER,1,  Voicemail(u${EXTENSION}|g(10))

and the volume remained the same.  I cannot find any good suggestions on
the internet in dealing with this issue.

  Can the volume of the recorded voice mail message be changed?  If so,
what I am doing wrong?  Any input would be greatly appreciated.  Thanks.


Best Regards,
John
  



Use video conversation to talk face-to-face with Windows Live Messenger.
Get started!
http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAG
LM_WL_Refresh_messenger_video_042008 

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Re: [asterisk-users] Spam: Voicemail- Recorded Mesage Low Volume

2008-04-01 Thread John Meksavan

When people leave me messages, both on the cellphone and POTS phones, on the 
recorded Asterisk voicemail message volume is really low.  I could barely hear 
my voicemail messages, when retrieving them, either cellphone or POTS line.

The voice mail prompts and sound recordings are fine, but the problem lies when 
people leave voicemails on the Asterisk box.  I could barely hear my voicemail 
messages.  

What can cause this?

Best Regards,
John

Date: Tue, 1 Apr 2008 14:46:40 -0500
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Spam: Voicemail- Recorded Mesage Low Volume

























Do you see the
same volume issues dialing in with a ‘normal’ POTS phone?

 

Are these the
standard recordings that come with Asterisk or some custom recordings?

 

…brig

 

Brig C. McCoy

ThyssenKrupp
Access Corp

Network
Administrator

Grandview, MO  64030

816-767-5577

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan

Sent: Tuesday, April 01, 2008 2:32
PM

To: Asterisk Users

Subject: Spam:[asterisk-users]
Voicemail- Recorded Mesage Low Volume



 

Asterisk Users,



  I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a
Debian Etch system.  On the recorded voice mail messages, the
volume is really low when retrieving them with my cell phone.  I tried
with multiple cell phones with the volume level high and still, the same
problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it
had no affect on the volume.  So, tried to add the g option on
the asterisk voicemail command, like this:



exten = s-NOANSWER,1,  Voicemail(u${EXTENSION}|g(10))



and the volume remained the same.  I cannot find any good suggestions on
the internet in dealing with this issue.



  Can the volume of the recorded voice mail message be changed?  If
so, what I am doing wrong?  Any input would be greatly appreciated. 
Thanks.





Best Regards,

John

  







Use video conversation to talk face-to-face with Windows
Live Messenger. Get started!







_
More immediate than e-mail? Get instant access with Windows Live Messenger.
http://www.windowslive.com/messenger/overview.html?ocid=TXT_TAGLM_WL_Refresh_instantaccess_042008___
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Re: [asterisk-users] [SPAM] - Re: Error checking asterisk method - suggestions? - Email found in subject

2008-02-14 Thread Johan Sandgren
 Hi there dear users and dear developers of Asterisk!


 I've got a maybe strange idea, let's see if you laugh or think it's 
 reasonable J

 I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering 
 machine).
 Each analog line can be reached through a phonenumber, since they are each 
 connected to my telephone provider. Yes expensive I know J


 The challenge:

 I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, 
 and 3) my operator is alive and working, and I have an Idea which I wonder 
 if will/could work.

 My first idea was to ask the zap-driver if it could detect if the line
 was ok, but no function existed to do that, what I could find. Anyone
 knows about one?

What do you mean by line is OK?

I mean, (being not so educated in the telephone technology), but to know that 
there is DC voltage connected to my ZAP-channel. (indicating status = OK)
According to Wiki, A calling party wishing to speak to another telephone will 
pick up the handset, thus operating the switch hook, which puts the telephone 
into active state or off hook with a resistance short across the wires, causing 
current to flow.
So, I suppose to know there is current flowing, would say I'm connected, but 
probably not guarantee that I can make calls. So this test would not give me 
trustworthy results. Or what do you say Tzafrir Cohen? (or others :)




 My second idea, was to try calling simply, to know if things were ok.
 But, I couldn't just call any number, I had to know the number was in
 use, and not disturbing anyone.
 So, I called myself, or I called another of my phonelines.

And you assume noone else calls in at the time?

Yes, since I only test lines, that haven't had any incoming calls for some 
time. Sure, someone COULD be using the line right when I'm trying to use it. 
How would
I know it was busy through AMI? Is it possible?


 So,
 I'd like to use the asterisk manager interface in java to originate a
 call from one ZAP-channel, calling out to my telephone provider,
 And then they will direct the call back to my, but into another
 ZAP-channel (since I'm calling that channel's number).

For a basic test that the line works, try TestClient and TestServer .
Originate a call from testclient (and set there the number. And set all
incoming calls temporarily to go to TestServer (did I mention the
assumption that noone calls in?)

One relatively cheap method of temporary is through setting a global
variable to a non-standard value. This means that the non-default value
will never last after a reload. And you can set the global even through
'core set global VARNAME VALUE' in the CLI.

Check the resulting reports in /var/lib/asterisk/testresults . Make sure
all of them were generated, and that none FAIL-ed.



I will look into this testserver  testclient. Totally new stuff to me :)
Hope I get it working.
Thanks for the tip Tzafrir !
Nice of you!



--
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icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] [Spam] Re: IAX softphone

2008-01-20 Thread Telecommunications

www.zoiper.com

We have been using this one, not like X-Lite but it works well.


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[asterisk-users] Spam Filter News

2007-11-01 Thread Thomas Kenyon
Is there any news on getting the Spam Filter fixed for this mailing list?

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Re: [asterisk-users] Spam Filter News

2007-11-01 Thread Jared Smith
On Thu, 2007-11-01 at 08:05 +, Thomas Kenyon wrote:
 Is there any news on getting the Spam Filter fixed for this mailing list?

We've tweaked it quite a bit in the past few weeks, and it seems to be
much better.  If you're still experiencing issues, please send me a
private email and I'll look into it in more detail.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Martin Smith
He changed the title of his response. Your post remains intact on the
list in its original form. In the interest of letting others decide, I
think it was a spammy post as well. Others have decided... Hence the
title of the list Asterisk Users Mailing List - Non-Commercial
Discussion. They decided it a long time ago. When you included pricing,
your email became commercial, an advertisement, and spam.

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 C. Savinovich
 Sent: Thursday, September 20, 2007 4:50 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: 
 Click-to-Call with VIDEO
 
 
 Please don't change the title of my post.  It is 
 disrespectful.  One thing
 is to give your opinion about its content, and another to be 
 self appointed
 editor of this forum.
 
 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even 
 longer if the
 community so demands it.  If you agree or disagree with it 
 fine, but let
 others decide.  They know spam when they see it.  Thank you.
 
 C. Savinovich
 VideoReps.net
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Anselm Martin
 Hoffmeister
 Sent: Thursday, September 20, 2007 10:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: 
 Click-to-Call with
 VIDEO
 
 Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
  Dear All:
  
  Just as the name suggests, and evolving from regular 
 Click-to-Call,
  Click-to-Call WITH VIDEO provides web sites with the 
 ability to engage
  their visitors with a live video agent (plus the phone 
 call).  All with
 just
  a click of a button placed on the customer's web site.  
 Please visit us at
  www.videoreps.net.
 
 When I read this, I thougt: Wow, here comes a nice, free, open,
 interesting software.
 
  Best of all, it is free for now!! (video only)...  Here 
 is the deal:
  Free for one month, no commitments.  Try, test it, call me. 
  After a month
  you can decide if you want to keep any of our plans:  
 Video-Call starts at
  $12.95 per month, and click-to-call with video at $29.95. 
 And yes, there
 is
  20% monthly commission if you include the service in the 
 pbxs you are
  selling.
 
 Well, not free at all.
 
  Note: Being an ActiveX component, please use internet explorer.
 
 And not having too much to do with open either, I guess.
 
 I would have called you to personally tell you that your mail was
 misplaced (there is some kind of asterisk-biz list, and I do 
 not read it
 for a purpose), but I do not use the integrated exploder for 
 lack of the
 necessary obfuscation system on my work machine.
 
 Please do not send commercials, ads and product information to this
 list. It might very well be considered SPAM. Just and only because
 _some_ readers might be interested there is no legitimation 
 for sending
 it (else every pen15-en1argm3nt _might_ trigger interest at some
 readers).
 
 Thanks
 Anselm
 
 
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Re: [asterisk-users] ***SPAM*** Announcing: Click-to -Call with VIDEO

2007-09-20 Thread Tilghman Lesher
On Thursday 20 September 2007, C. Savinovich wrote:
 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even longer
 if the community so demands it.  If you agree or disagree with it
 fine, but let others decide.  They know spam when they see it.  Thank
 you.

I am one of the longest running developers of Asterisk and I agree with
Anselm's assessment.  Please post announcements like this in the future
to the asterisk-biz list, as that list is for business discussions.
This list is for users of Asterisk, not for commercial announcements.

-- 
Tilghman

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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Dean Collins
I was interested in it - commercial or otherwise.but only because I
used to work for the competition.

Commercial or otherwise it looks like a very cool technology and
something I'd be interested in - but only as a one of purchase price
rather than an ASP.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Martin Smith
 Sent: Thursday, 20 September 2007 1:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call
with VIDEO
 
 He changed the title of his response. Your post remains intact on the
 list in its original form. In the interest of letting others decide, I
 think it was a spammy post as well. Others have decided... Hence the
 title of the list Asterisk Users Mailing List - Non-Commercial
 Discussion. They decided it a long time ago. When you included
pricing,
 your email became commercial, an advertisement, and spam.
 
 Martin Smith, Systems Developer
 [EMAIL PROTECTED]
 Bureau of Economic and Business Research
 University of Florida
 (352) 392-0171 Ext. 221
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  C. Savinovich
  Sent: Thursday, September 20, 2007 4:50 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] ***SPAM*** Announcing:
  Click-to-Call with VIDEO
 
 
  Please don't change the title of my post.  It is
  disrespectful.  One thing
  is to give your opinion about its content, and another to be
  self appointed
  editor of this forum.
 
  We posted in this forum because it is a contribution to the asterisk
  community, and because it is free for a month, and maybe even
  longer if the
  community so demands it.  If you agree or disagree with it
  fine, but let
  others decide.  They know spam when they see it.  Thank you.
 
  C. Savinovich
  VideoReps.net
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Anselm Martin
  Hoffmeister
  Sent: Thursday, September 20, 2007 10:01 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] ***SPAM*** Announcing:
  Click-to-Call with
  VIDEO
 
  Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
   Dear All:
  
   Just as the name suggests, and evolving from regular
  Click-to-Call,
   Click-to-Call WITH VIDEO provides web sites with the
  ability to engage
   their visitors with a live video agent (plus the phone
  call).  All with
  just
   a click of a button placed on the customer's web site.
  Please visit us at
   www.videoreps.net.
 
  When I read this, I thougt: Wow, here comes a nice, free, open,
  interesting software.
 
   Best of all, it is free for now!! (video only)...  Here
  is the deal:
   Free for one month, no commitments.  Try, test it, call me.
   After a month
   you can decide if you want to keep any of our plans:
  Video-Call starts at
   $12.95 per month, and click-to-call with video at $29.95.
  And yes, there
  is
   20% monthly commission if you include the service in the
  pbxs you are
   selling.
 
  Well, not free at all.
 
   Note: Being an ActiveX component, please use internet explorer.
 
  And not having too much to do with open either, I guess.
 
  I would have called you to personally tell you that your mail was
  misplaced (there is some kind of asterisk-biz list, and I do
  not read it
  for a purpose), but I do not use the integrated exploder for
  lack of the
  necessary obfuscation system on my work machine.
 
  Please do not send commercials, ads and product information to this
  list. It might very well be considered SPAM. Just and only because
  _some_ readers might be interested there is no legitimation
  for sending
  it (else every pen15-en1argm3nt _might_ trigger interest at some
  readers).
 
  Thanks
  Anselm
 
 
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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Troy Ayers
snip
 Please don't change the title of my post.  It is 
 disrespectful.  One thing
 is to give your opinion about its content, and another to be 
 self appointed
 editor of this forum.

 We posted in this forum because it is a contribution to the asterisk
 community, and because it is free for a month, and maybe even 
 longer if the
 community so demands it.  If you agree or disagree with it 
 fine, but let
 others decide.  They know spam when they see it.  Thank you.

 C. Savinovich
 VideoReps.net
snip

As one of the others I say it looks like spam to me too.  I won't be 
trying your product anytime soon, partly because of the way the matter 
was handled.


-Troy


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Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO

2007-09-20 Thread Steve Totaro
Speaking for the Asterisk community as a whole, we demand that it be 
free forever.  Please honor your statement below, We posted in this 
forum because it is a contribution to the asterisk
  community, and because it is free for a month, and maybe even longer 
if the
  community so demands it.



C. Savinovich wrote:
 Please don't change the title of my post.  It is disrespectful.  One thing
 is to give your opinion about its content, and another to be self appointed
 editor of this forum.
 
   If you agree or disagree with it fine, but let
 others decide.  They know spam when they see it.  Thank you.
 
 C. Savinovich
 VideoReps.net
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin
 Hoffmeister
 Sent: Thursday, September 20, 2007 10:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with
 VIDEO
 
 Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich:
 Dear All:

 Just as the name suggests, and evolving from regular Click-to-Call,
 Click-to-Call WITH VIDEO provides web sites with the ability to engage
 their visitors with a live video agent (plus the phone call).  All with
 just
 a click of a button placed on the customer's web site.  Please visit us at
 www.videoreps.net.
 
 When I read this, I thougt: Wow, here comes a nice, free, open,
 interesting software.
 
 Best of all, it is free for now!! (video only)...  Here is the deal:
 Free for one month, no commitments.  Try, test it, call me.  After a month
 you can decide if you want to keep any of our plans:  Video-Call starts at
 $12.95 per month, and click-to-call with video at $29.95. And yes, there
 is
 20% monthly commission if you include the service in the pbxs you are
 selling.
 
 Well, not free at all.
 
 Note: Being an ActiveX component, please use internet explorer.
 
 And not having too much to do with open either, I guess.
 
 I would have called you to personally tell you that your mail was
 misplaced (there is some kind of asterisk-biz list, and I do not read it
 for a purpose), but I do not use the integrated exploder for lack of the
 necessary obfuscation system on my work machine.
 
 Please do not send commercials, ads and product information to this
 list. It might very well be considered SPAM. Just and only because
 _some_ readers might be interested there is no legitimation for sending
 it (else every pen15-en1argm3nt _might_ trigger interest at some
 readers).
 
 Thanks
 Anselm
 


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[asterisk-users] Spam filter woes [was: Astricon 2007 -- does anybody need a ride?]

2007-09-17 Thread Jared Smith
On Mon, 2007-09-17 at 09:51 -0600, Stephen Bosch wrote:
 I posted it at least four times, and not one made it through. Perhaps
 it's a spam filter.

Yeah, I'm looking into the problem now.  It seems to be related to the
spam filter sitting in front of the mailing list server.  I'll post an
update when I know more. 

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[Asterisk-Users] Spam

2006-05-09 Thread Michael Welter
I was just spammed from exoxshaindia.com.  The spoofed from address was 
asterisk-users asterisk-users@lists.digium.com, and the subject was 
Fw: Real show and it containd attachment 3.92315089702606R02.UUE.  I 
believe UUE is a compressed executable.


WATCH OUT!


--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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[Asterisk-Users] spam filter

2005-06-03 Thread Amicus Curious

Fixed it! Try these keywords:
Viagra
Viiagra
V I A G R A
V_I_A_G_R_A
V_E_S_T_E_R_L_I_N_G
Karl


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[Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM

2005-04-20 Thread Al
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.

I tried several times - obviously unsuccessfully, as you can see - to
unsubscribe from this list, and sice that did not work, i set my mail
server to BOUNCE list messages.

But nope, nobody gets it - the mail server is deaf and dumb and keeps
sending me dozens and dozens of messages.

If there is anybody reading this who can reach the person(s) in charge of
the mailing list server, please tell them that they are having (and
causing) a problem. ;-)

Al

--


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Re: [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM

2005-04-20 Thread Jean-Michel Hiver
Al wrote:
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.
 

You receive messages from this list because YOU signed up for it.
If you knew anything about email (or mailing lists), you could have 
looked in the email headers to find the unsubscribe address:

http://lists.digium.com/mailman/listinfo/asterisk-users

I tried several times - obviously unsuccessfully, as you can see - to
unsubscribe from this list, and sice that did not work, i set my mail
server to BOUNCE list messages.
 

Oh great. Somebody with their own mailserver who doesn't know email, 
mail filters or mailing lists.

But nope, nobody gets it - the mail server is deaf and dumb and keeps
sending me dozens and dozens of messages.
 

Which in the worst case scenario (unsubscribe don't work! - doubtful) 
you could filter out or simply drop at the mail server level.

If there is anybody reading this who can reach the person(s) in charge of
the mailing list server, please tell them that they are having (and
causing) a problem. ;-)
 

Somehow I think you're your own problem...
Cheers,
Jean-Michel.
--
Ykoz Un Max - La VoIP en pré-payé!
Essayez gratuitement - 5 crédits offerts.
--- http://ykoz.net/voip/max ---
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Re: [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM

2005-04-20 Thread Ronald Wiplinger
Al wrote:
Yeah... unbelieveable but true: spam, defined often us undesired bulk
mail comes in many forms, including messages from this list.
I tried several times - obviously unsuccessfully, as you can see - to
unsubscribe from this list, and sice that did not work, i set my mail
server to BOUNCE list messages.
But nope, nobody gets it - the mail server is deaf and dumb and keeps
sending me dozens and dozens of messages.
If there is anybody reading this who can reach the person(s) in charge of
the mailing list server, please tell them that they are having (and
causing) a problem. ;-)
Al
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Have you tried to read and follow the last line of each message?
--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com+886 (0) 939--77-55-16  or FWD 511208
- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org
PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.

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Re: [Asterisk-Users] SPAM SPAM SPAM SAM SPAM SPAM SPAM

2005-04-20 Thread C F
On 4/20/05, Al [EMAIL PROTECTED] wrote:
 Yeah... unbelieveable but true: spam, defined often us undesired bulk
 mail comes in many forms, including messages from this list.
 
 I tried several times - obviously unsuccessfully, as you can see - to
 unsubscribe from this list,

Obviously the problem is you, because it works for all of us.

 and sice that did not work, i set my mail
 server to BOUNCE list messages.

If you set your server to bounce it, how do you still get it? do you
check in the queue to make sure that it's archived as bad mail?


 
 But nope, nobody gets it - the mail server is deaf and dumb and keeps
 sending me dozens and dozens of messages.
 
 If there is anybody reading this who can reach the person(s) in charge of
 the mailing list server, please tell them that they are having (and
 causing) a problem. ;-)

I think it's you that is having a problem, eat breadfast and then try again.

 
 Al
 
 --
 
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[Asterisk-Users] Spam: I really need help with this!!!!!!

2004-11-18 Thread Rodney Acosta Coya
when i try to compile asterisk-1.0.2 i get this error :

configure: error: termcap support not found

how i can install this suport.

for more information:

linux:/inst/pbx/asterisk-1.0.2 # make
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C
 $x depend || exit 1 ; done
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/stdtime'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/stdtime'
cd editline  unset CFLAGS LIBS  test -f config.h || ./configure
loading cache ./config.cache
checking for gcc... gcc
checking whether the C compiler (gcc  ) works... yes
checking whether the C compiler (gcc  ) is a cross-compiler... no
checking whether we are using GNU C... yes
checking whether gcc accepts -g... yes
checking how to run the C preprocessor... gcc -E
checking host system type... i686-pc-linux-gnu
checking for a BSD compatible install... install
checking for ranlib... ranlib
checking for ar... /usr/bin/ar
checking for tgetent in -ltermcap... no
checking for tgetent in -ltinfo... no
checking for tgetent in -lcurses... no
checking for tgetent in -lncurses... no
configure: error: termcap support not found
make: *** [editline/libedit.a] Error 1
linux:/inst/pbx/asterisk-1.0.2 # make install
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C
 $x depend || exit 1 ; done
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/stdtime'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/stdtime'
cd editline  unset CFLAGS LIBS  test -f config.h || ./configure
loading cache ./config.cache
checking for gcc... gcc
checking whether the C compiler (gcc  ) works... yes
checking whether the C compiler (gcc  ) is a cross-compiler... no
checking whether we are using GNU C... yes
checking whether gcc accepts -g... yes
checking how to run the C preprocessor... gcc -E
checking host system type... i686-pc-linux-gnu
checking for a BSD compatible install... install
checking for ranlib... ranlib
checking for ar... 

Re: [Asterisk-Users] Spam: I really need help with this!!!!!!

2004-11-18 Thread Sean Kennedy
Rodney Acosta Coya wrote:
when i try to compile asterisk-1.0.2 i get this error :
configure: error: termcap support not found
how i can install this suport.
for more information:
linux:/inst/pbx/asterisk-1.0.2 # make
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C
$x depend || exit 1 ; done
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/stdtime'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/stdtime'
cd editline  unset CFLAGS LIBS  test -f config.h || ./configure
loading cache ./config.cache
checking for gcc... gcc
checking whether the C compiler (gcc  ) works... yes
checking whether the C compiler (gcc  ) is a cross-compiler... no
checking whether we are using GNU C... yes
checking whether gcc accepts -g... yes
checking how to run the C preprocessor... gcc -E
checking host system type... i686-pc-linux-gnu
checking for a BSD compatible install... install
checking for ranlib... ranlib
checking for ar... /usr/bin/ar
checking for tgetent in -ltermcap... no
checking for tgetent in -ltinfo... no
checking for tgetent in -lcurses... no
checking for tgetent in -lncurses... no
configure: error: termcap support not found
make: *** [editline/libedit.a] Error 1
linux:/inst/pbx/asterisk-1.0.2 # make install
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C
$x depend || exit 1 ; done
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/res'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/channels'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/pbx'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/apps'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/codecs'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/formats'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/agi'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/cdr'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/astman'
make[1]: Entering directory `/inst/pbx/asterisk-1.0.2/stdtime'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/inst/pbx/asterisk-1.0.2/stdtime'
cd editline  unset CFLAGS LIBS  test -f config.h || ./configure
loading cache ./config.cache
checking for gcc... gcc
checking whether the C compiler (gcc  ) works... yes
checking whether the C compiler (gcc  ) is a cross-compiler... no
checking whether we are using GNU C... yes
checking whether gcc accepts -g... yes
checking how to run the C preprocessor... gcc -E
checking host system type... i686-pc-linux-gnu
checking for a BSD compatible install... install
checking for ranlib... ranlib

Re: [Asterisk-Users] Spam: I really need help with this!!!!!!

2004-11-18 Thread Steven Critchfield
On Thu, 2004-11-18 at 16:26 -0500, Rodney Acosta Coya wrote:
 when i try to compile asterisk-1.0.2 i get this error :
 
 configure: error: termcap support not found
 
 how i can install this suport.

Thats a linux distribution question, not an asterisk question. Look for
a termcap package and don't forget to get the -dev or -devel package as
well.
-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] Spam: I really need help with this!!!!!!

2004-11-18 Thread Peter Hoppe
I had the same problem when I installed Asterisk (it runs on SuSE Linux 
9.0 now). You can find info in this mailing list under

http://lists.digium.com/pipermail/asterisk-users/2003-May/011185.html
(hint hint) Some Googling helps... thats how I found this link :)
--
There are 10 kinds of people in the world,
those who understand binary, and those who don't.

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Re: [Asterisk-Users] SPAM Notice

2004-10-19 Thread Benjamin on Asterisk Mailing Lists
On Tue, 19 Oct 2004 10:34:02 +1000, Adam Goryachev
[EMAIL PROTECTED] wrote:
 Just a heads-up that asterisk is getting a mention in spam now... oh,
 and make sure you NEVER EVER buy anything from this company.
 
 [SNIP]
 NEWS: VocalScape Inc. Announces DELETED for Asterisk IP PBX Users.

As marketeers say: Any news is good news. So, this may well be good news.

Just make sure you have your mailboxes protected or use a webmail
account to participate in the mailing lists and be prepared for the
wave of Asterisk related spam that might be coming our way. Thanks for
the warning.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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[Asterisk-Users] SPAM Notice

2004-10-18 Thread Adam Goryachev
Just a heads-up that asterisk is getting a mention in spam now... oh,
and make sure you NEVER EVER buy anything from this company.

[SNIP]
NEWS: VocalScape Inc. Announces DELETED for Asterisk IP PBX Users. You
can start earning money NOW ? without waiting for the fabled
stock-market recovery. How? By
[END]

BTW, this was sent to a closed, announce only mailing list, which for
some reason seems to be on some sort of spam database because I get a
large number of spams daily which I need to reject.

Regards,
Adam



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