Re: [asterisk-users] AOC-E pass through

[Olivier]

2009/2/4 Jean-Denis Girard jd.gir...@sysnux.pf

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Klaus Darilion a écrit :
  Take a look at http://bugs.digium.com/view.php?id=7494

 Thanks for the pointer; I'm already monitoring this issue, but there
 seems to be no progress on that, unfortunately.

 
  Unfortunately it is not yet included in Asterisk, as the patch is
  somehow a workaround (e.g. faking AOC-E based on last AOC-D).

 Here the telco is not sending AOC-D, just AOC-E.


just for curiosity, is AOC-E messages sending included in telco basic
subscription or is an option needed for that ?
cheers



 
  Nevertheless a customer of us uses it for some years now (Astersik 1.2)
  without any problems.
 
  regards
  klaus
 
  Jean-Denis Girard schrieb:
  Hi,
 
  I'd like to know what is the current situation with regard to AOC-E,
  when Asterisk is inserted between the telco and an existing PBX, using
  E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the
  telco to the PBX, so that billing system still works? The system would
  be for a hotel, so breaking billing system is not possible.
 


 Thanks,
 - --
 Jean-Denis Girard

 SysNux  Systèmes  Linux  en Polynésie française
 http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
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[asterisk-users] A Simple Asterisk Based Toll Fraud Prevention Script

[J. Oquendo]

Subject says it all...

A Simple Asterisk Based Toll Fraud Prevention Script
http://www.infiltrated.net/asterisk-ips.html

Ramblings for admins/engineers to think about. Doesn't
have to cost you umteen thousand dollars for stuff like
IPS/IDS. Although a little on the crude side, quite
effective. If you care to dabble with MySQL you can
create quite an impressive hosts based IPS that is
custom tailored to your infrastructure.

Anyhow, was bored (ADHD) and wanted to ramble on for
a little while.


=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP

Enough research will tend to support your
conclusions. - Arthur Bloch

A conclusion is the place where you got
tired of thinking - Arthur Bloch

227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E


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Re: [asterisk-users] Security issue

[David fire]

you have many options but you should use it together.
firewall

in the user/peers definitions add host=ip
and/or
deny=0.0.0.0/0.0.0.0
permit=ip/mask

change the ip of your server.

use something like ossec to avoid force brute.

David

2009/2/6 oumar ndiaye ond4...@gmail.com

 Is there a way to restrict connection to my asterisk server to users based
 on their IP addresses, and not just password. I have some hackers who
 connect to my server to make illegitimate solicitation calls to people. I
 had to shutdown the server for now until I find a solution. ANY HELP?
  Thanks.
 ond

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Re: [asterisk-users] Asterisk 3rd party developed commercial software sales licensing platform

[Dean Collins]

I think it's time to 'ping' John Todd and Digium on this topic again.

 

What happened Digium?

 

Why did you say you were going to take this project on but have not come
back to the community with an answer yet?

 

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

 





From: Dean Collins 
Sent: Sunday, 4 May 2008 3:45 PM
Subject: Asterisk 3rd party developed commercial software sales
licensing platform

 

 

Hi Randy,

 

As discussed on Friday the 9th of May I would like to host the
Voip Users Conference Call.

 

The purpose of this call is to discuss the community's feelings
about an Asterisk 3rd party developed commercial software sales
licensing platform.

 

The plan is that some form of documented published schema be
implemented that will allow for 3rd party software developers to sell
their software applications using a common licensing model similar to
the way G729 licenses are sold by Digium. 

 

 

 

Basically this discussion came about for a 3rd party ecosystem
question a few weeks ago when Cory Andrews from VoIP supply was on the
Voip-Users conference call.

I asked the question - how much of VoIP Supply revenue is
product hardware versus applications - he said we don't sell any
services such as ITSP hosted Asterisk so I replied that wasn't what I
was thinking of and gave the example of Snap Dialer which is a low cost
(I paid $20 for it) application which allows me to dial names from
Outlook.

 

He said they didn't sell any applications like this at all but
would consider selling them if this was an opportunity presented to him.

I then talked about some of the consulting I did for
Salesforce.com and how they have built an entire ecosystem of third
party applications all built by other people apart from salesforce.com
but utilizing the documented API's and application security /licensing
etc.

My comments were that although Asterisk should always remain a
free open source application that developers need to eat and pay rent as
well.

If there was some common marketplace that developers could sell
small - low cost third party applications to the Asterisk community that
Digium had some type of overview/management control over who listed etc
that this would deliver a stream of revenue that would encourage further
application development.

The question I then posed to the group was if anyone knew how
Digium managed the sale and licensing of the G729 codes.
And if this was an open published standard that could it be used
as the basis for the Asterisk ecosystem license model.

Now I know it's not perfect and can be hacked but everything can
be hacked. The idea is to build apps cheap enough that it's not worth
the effort of hacking. If anyone has some alternative suggestions on how
apps should be licensed we'd like to hear them this Friday.

I know there were discussions in the early days of the Mexuar
launch about how they could license a single channel of the Mexuar
Corraleta application rather than the entire server license for $2000.
The issue always came down to how we could license it to 1/ a single
channel license. 2/ tied to a single machine and not transferable
(currently the Mexuar license is hard coded in the application to the
servers IP address).

 

I know for me personally although I have donated to numerous
bounty requests (I even tried to get one developed for video
conferencing a few years ago that was around the $10,000 range) I
haven't seen the ongoing continual development that would benefit the
Asterisk community.

 

*   I personally would be more than happy to pay for 'the
next generation of FOP', it was a great application when launched but
there is a lot more it could be offering. 

 

*   I'd also like to implement a far smarter 'user
dashboard' similar to what Druid are developing. 

 

*   Now I no longer work for Mexuar and don't have access to
it anymore I'd also like to pay for a single channel Mexuar license
rather than using 'lesser quality' experiences by other solutions. 

 

*   Drawing on my own now defunct project - is the Asterisk
user community now ready for centrally provided services such as the
'off-deck processing' like the Tellme Speech Recognition Service
http://www.voip-info.org/wiki/view/Tellme . As demonstrated by Amazon
EC2 / S3 web services I'm a huge fan of cloud computing off-deck
processing, Should these style services also be able to take advantage
of an Asterisk 3rd party ecosystem licensing model. 

 

[asterisk-users] GROUP() decrement

[Thomas Winter]

Hi,

how I can decrement the value of GROUP_COUNT() by one after I have before used 
GROUP(), so that other channel will get the correct value of GROUP_COUNT().


for examaple

exten = _X!,n,Set(GROUP()=${Provider})
exten = _X!,n,DIAL(SIP/${ext...@${provider})

When Dialstatus is CONGESTION I want to dial again with another provider but I 
have to decrement the GROUP of the unused provider.

If there is no function, woud it be possible to call GROUP() in the Macro 
called by the DIAL command if DIALSTATUS is ANSWERED?

Or do I have to progamm it outside with AGI?

best regards
Thomas





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[asterisk-users] One way audio after IVR tree

[James Lamanna]

Hi,
I have a couple of users who are having a peculiar problem.
On some outbound numbers where there is a deep IVR tree (3+
selections), and then a live person picks up,
the live person will be unable to hear them on the phone, but they can
hear the live person.
I've done packet traces and it appears as though audio is being passed
both ways, but the audio
from the caller is severely muted before it gets to asterisk.

Has anyone seen this before? It's almost like the phone thinks its
still sending DTMF or something
and mutes the audio. I've seen this happen on both linksys 942 and 962 phones.

Thanks.

-- James

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Re: [asterisk-users] VPN and Asterisk

[Dave Platt]

 One of my user was asking, can he use VPN to access asterisk ?
 What does it mean ?

 And its possible ?

 How ?VPN

Yes, it's possible.

As one example: I have the OpenVPN software installed on my Asterisk
server, and on my Nokia N810 wireless Internet tablet.  The tablet is
configured to use the VPN's server-side IP address as its SIP server.
A similar sort of system could be set up with other VPN packages (e.g.
CIPE, Cisco's offerings, etc.),

This approach has several advantages, compared to the alternative (not
using a VPN, and turning on STUN support in the client):

-  All of the SIP and RTP traffic to/from the tablet is encrypted, and
   thus relatively resistant to evesdropping.
   
-  The tablet and the Asterisk server have IP addresses for each other
   which are being established by the VPN software, and don't need to
   be (and aren't) altered or translated by access-point or corporate
   routers.  This pretty much eliminates the common I can't get audio
   in one or both directions problem with using SIP through private
   IP networks and NAT routers.
   
-  Most network firewalls will pass VPN traffic, even if they haven't
   been configured to pass raw SIP and RTP.
   
There are some disadvantages, though:

-  Some amount of CPU overhead at both ends, and perhaps some
   increased latency (the latter is minor, I believe).
   
-  The RTP traffic must flow through the VPN/Asterisk server... it
   cannot be reinvited into a direct connection between the tablet
   and the destination, because the tablet is using an IP address for
   the connection which exists only on the VPN and isn't externally
   reachable.

This sort of VPN setup (where the Asterisk client is on the same
system that's running the VPN software) is the one you'd want to use
for many road warrior setups.

VPNs can also be used to set up secure IP tunnels between two
different, remotely-located networks.  This might be done to tie
together (e.g.) two different offices, each having its own Asterisk
server.


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Re: [asterisk-users] VPN and Asterisk

[Bill Michaelson]

 David @ULC ucoms2...@gmail.com wrote

One of my user was asking, can he use VPN to access asterisk ?
What does it mean ?

And its possible ?

How ?VPN


Sometimes what is called a VPN is not a VPN by everyone's definition, so 
beware. By my definition, a (IP) VPN supports full layer 3 functionality 
(and sometimes more), as opposed to, say, some type of proxy that relays 
a limited set of protocols over a particular path with encryption. So 
you need to be more specific about your question.


Example:

I've run Asterisk over OpenVPN. In this case, no problem; it's just 
another networking layer and the only special consideration is increased 
overhead, and that same consideration applies to any application. There 
are other VPNs.





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[asterisk-users] Minimum version for asterisk and iaxmodem

[James Lamanna]

Hi,
I'm trying to use iaxmodem against a very old version of asterisk
(1.0.7 - its a debian sarge embedded system),
yet when asterisk gets a call from iaxmodem, it says that the format
for the call is unknown.
Does anyone know if there is a minimum version of asterisk that is
compatible with iaxmodem 1.1.0?

Thanks.

-- James

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Re: [asterisk-users] Minimum version for asterisk and iaxmodem

[Lee Howard]

James Lamanna wrote:
 Hi,
 I'm trying to use iaxmodem against a very old version of asterisk
 (1.0.7 - its a debian sarge embedded system),
 yet when asterisk gets a call from iaxmodem, it says that the format
 for the call is unknown.
 Does anyone know if there is a minimum version of asterisk that is
 compatible with iaxmodem 1.1.0?

I originally developed IAXmodem while using Asterisk 1.2.x.  I have 
since migrated to Asterisk 1.4.x.  I never attempted to use Asterisk 
1.0.x with IAXmodem, and I have also never tried Asterisk 1.6.x 
(although I suspect others have without issue).

Thanks,

Lee.

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Re: [asterisk-users] Minimum version for asterisk and iaxmodem

[James Lamanna]

On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote:
 James Lamanna wrote:

 Hi,
 I'm trying to use iaxmodem against a very old version of asterisk
 (1.0.7 - its a debian sarge embedded system),
 yet when asterisk gets a call from iaxmodem, it says that the format
 for the call is unknown.
 Does anyone know if there is a minimum version of asterisk that is
 compatible with iaxmodem 1.1.0?

 I originally developed IAXmodem while using Asterisk 1.2.x.  I have since
 migrated to Asterisk 1.4.x.  I never attempted to use Asterisk 1.0.x with
 IAXmodem, and I have also never tried Asterisk 1.6.x (although I suspect
 others have without issue).

Thanks Lee.
I'm compiling 1.2.31 right now to see if I have more luck (1.4.x is
giving me trouble w/
iaxmodem + hylafax on this ARM platform, but it works great on x86!).


 Thanks,

 Lee.


-- James

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Re: [asterisk-users] Minimum version for asterisk and iaxmodem

[James Lamanna]

So I got 1.2.31 compiled but I still can't do a loopback fax with
IAXModem, Hylafax and Asterisk.
The call connects, but the iaxmodem log fills up with Adjusting skew
xxx and the fax ultimately
fails with:

Feb 07 14:10:26.05: [ 3497]: MODEM No carrier
Feb 07 14:10:26.05: [ 3497]: Failure to receive silence
(synchronization failure). {E100}
Feb 07 14:10:26.05: [ 3497]: RECV FAX: Failure to receive silence
(synchronization failure). {E100}
Feb 07 14:10:26.05: [ 3497]: RECV FAX: end
Feb 07 14:10:26.05: [ 3497]: Failure to receive silence
(synchronization failure). {E100}
Feb 07 14:10:26.05: [ 3497]: SESSION END

 -- Accepting AUTHENTICATED call from 127.0.0.1:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing Dial(IAX2/iaxmodem0-11201,
IAX2/iaxmodem1/xx|60) in new stack
-- Called iaxmodem1/6266053472
-- Call accepted by 127.0.0.1 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxmodem1-1796 is ringing
-- IAX2/iaxmodem1-1796 answered IAX2/iaxmodem0-11201
-- Attempting native bridge of IAX2/iaxmodem0-11201 and IAX2/iaxmodem1-1796




On Sat, Feb 7, 2009 at 1:44 PM, James Lamanna jlama...@gmail.com wrote:
 On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote:
 James Lamanna wrote:

 Hi,
 I'm trying to use iaxmodem against a very old version of asterisk
 (1.0.7 - its a debian sarge embedded system),
 yet when asterisk gets a call from iaxmodem, it says that the format
 for the call is unknown.
 Does anyone know if there is a minimum version of asterisk that is
 compatible with iaxmodem 1.1.0?

 I originally developed IAXmodem while using Asterisk 1.2.x.  I have since
 migrated to Asterisk 1.4.x.  I never attempted to use Asterisk 1.0.x with
 IAXmodem, and I have also never tried Asterisk 1.6.x (although I suspect
 others have without issue).

 Thanks Lee.
 I'm compiling 1.2.31 right now to see if I have more luck (1.4.x is
 giving me trouble w/
 iaxmodem + hylafax on this ARM platform, but it works great on x86!).


 Thanks,

 Lee.


 -- James


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[asterisk-users] can anybody tell me how Magic jack can be so cheap ????

[Ignacio Ortega A.]

How Magic Jack can only charge $20 per year?

do they have a call limit?
do they have a call duration limit or limit of minutes per day?,


Thanks
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Re: [asterisk-users] Minimum version for asterisk and iaxmodem

[Tzafrir Cohen]

On Sat, Feb 07, 2009 at 01:08:32PM -0800, James Lamanna wrote:
 Hi,
 I'm trying to use iaxmodem against a very old version of asterisk
 (1.0.7 - its a debian sarge embedded system),

You can find some later backports of Asterisk to Sarge at the (now
disfunctional) http://pkg-voip.buildserver.net/ .

1.2.13 was definetly built on Sarge. Likewise later 1.2-s . I figure you
can get 1.4 to build there as well by playing a bit with build
dependencies and such.

-- 
   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] Security issue

[oumar ndiaye]

David,
Thanks in advance. Where do I change the user/peers definition? Is it in the
firewall of the OS? In that case that won't work because the server host
other services such as ssh http that are open to any IP as long as the user
has the correct credentials. Doesn't asterisk itself has built in security
filters?

If the only choice is to do in the OS's firewall, then I will need to
include the port numbers of SIP, IAX in my firewall rules. In this case,
which ports should I block to keep unwanted SIP/IAX connections from
specific IP's.
Thanks.

On Sat, Feb 7, 2009 at 9:29 AM, David fire ddf...@gmail.com wrote:

 you have many options but you should use it together.
 firewall

 in the user/peers definitions add host=ip
 and/or
 deny=0.0.0.0/0.0.0.0
 permit=ip/mask

 change the ip of your server.

 use something like ossec to avoid force brute.

 David

 2009/2/6 oumar ndiaye ond4...@gmail.com

  Is there a way to restrict connection to my asterisk server to users
 based on their IP addresses, and not just password. I have some hackers who
 connect to my server to make illegitimate solicitation calls to people. I
 had to shutdown the server for now until I find a solution. ANY HELP?
 Thanks.
 ond

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 (\__/)
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-- 
Oumar Ndiaye
CTO
ANTG Telecom
www.antg.com
ondi...@antg.com
ondi...@alum.mit.edu
ond4...@gmail.com
Tel: +1-919-291-8742
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Re: [asterisk-users] Security issue

[Eric Fort]

use IP tables and start with deny all.  Follow this by allowing only
the protocols/ports you want and only the source/destination ip's you
wish to allow.  these can be combined to say allow ssh from anywhere
but only allow sip (and it's range of ports) to/from a very limited
set of ip's belonging to say your ITSP.  for users that move about a
bunch they can use vpn to an allowed subnet.

Eric

On Sat, Feb 7, 2009 at 5:47 PM, oumar ndiaye ondi...@antg.com wrote:
 David,
 Thanks in advance. Where do I change the user/peers definition? Is it in the
 firewall of the OS? In that case that won't work because the server host
 other services such as ssh http that are open to any IP as long as the user
 has the correct credentials. Doesn't asterisk itself has built in security
 filters?

 If the only choice is to do in the OS's firewall, then I will need to
 include the port numbers of SIP, IAX in my firewall rules. In this case,
 which ports should I block to keep unwanted SIP/IAX connections from
 specific IP's.
 Thanks.

 On Sat, Feb 7, 2009 at 9:29 AM, David fire ddf...@gmail.com wrote:

 you have many options but you should use it together.
 firewall

 in the user/peers definitions add host=ip
 and/or
 deny=0.0.0.0/0.0.0.0
 permit=ip/mask

 change the ip of your server.

 use something like ossec to avoid force brute.

 David

 2009/2/6 oumar ndiaye ond4...@gmail.com

 Is there a way to restrict connection to my asterisk server to users
 based on their IP addresses, and not just password. I have some hackers who
 connect to my server to make illegitimate solicitation calls to people. I
 had to shutdown the server for now until I find a solution. ANY HELP?
 Thanks.
 ond
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 --
 Oumar Ndiaye
 CTO
 ANTG Telecom
 www.antg.com
 ondi...@antg.com
 ondi...@alum.mit.edu
 ond4...@gmail.com
 Tel: +1-919-291-8742


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Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????

[C F]

They believe they have advertisement revenues.

On Sat, Feb 7, 2009 at 5:45 PM, Ignacio Ortega A. nachom...@gmail.com wrote:
 How Magic Jack can only charge $20 per year?

 do they have a call limit?
 do they have a call duration limit or limit of minutes per day?,


 Thanks

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Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????

[Forrest W Christian]

Or more accurately, they believe they can follow the NetZero or Juno 
model (Free in exchange for ads being pushed to you).

-forrest

C F wrote:
 They believe they have advertisement revenues.

 On Sat, Feb 7, 2009 at 5:45 PM, Ignacio Ortega A. nachom...@gmail.com wrote:
   
 How Magic Jack can only charge $20 per year?

 do they have a call limit?
 do they have a call duration limit or limit of minutes per day?,


 Thanks

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Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????

[k4rjj]

 

Never seen any ads except for them. Actually the thing sits on a server down in the garage so I don't see anything! Darn thing just works! I bought it as a second line when the wife is using the copper line to work.Ronny K4RJJ

 -- Original message from Forrest W Christian f...@mt.net: --


 Or more accurately, they believe they can follow the NetZero or Juno 
 model (Free in exchange for ads being pushed to you).
 
 -forrest
 
 C F wrote:
  They believe they have advertisement revenues.
 
  On Sat, Feb 7, 2009 at 5:45 PM, Ignacio Ortega A.  wrote:

  How Magic Jack can only charge $20 per year?
 
  do they have a call limit?
  do they have a call duration limit or limit of minutes per day?,
 
 
  Thanks
 
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Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????

[Sam]

k4...@bellsouth.net wrote:
 
  Never seen any ads except for them.  Actually the thing sits on a 
 server down in the garage so I don't see anything!  Darn thing just 
 works!  I bought it as a second line when the wife is using the copper 
 line to work.
 
 
 Ronny K4RJJ
 


The way most people interpret the eula is that magic jack can actually 
listen in on your calls and send you customized ads based on what you 
talk about.  How you get these ads I don't know.  I am also too lazy to 
read the eula myself, and I don't use the service anyway. 
http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html

Sam

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Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????

[Kinjal Dixit]

the ads will start once there is critical mass.  the following are the
scenarios for ads:
1. when you dial a number, before hearing the ringing, you have to listen to
an ad.  the length of the ad would be proportional to the intensity of your
usage... the more you use, the longer the ads.
2. when the caller answers, they will first hear magic jack promo, then they
will hear your voice.
3. the call in interrupted every few minutes to play an ad to both parties.
4. they will give an ad free service if you pay a higher charge.

I just hope I am not giving the people at magicjack any ideas, but if I am,
I would sure appreciate if they pay me!!


On Sun, Feb 8, 2009 at 9:58 AM, k4...@bellsouth.net wrote:

   Never seen any ads except for them.  Actually the thing sits on a server
 down in the garage so I don't see anything!  Darn thing just works!  I
 bought it as a second line when the wife is using the copper line to work.


 Ronny K4RJJ

 -- Original message from Forrest W Christian f...@mt.net:
 --


  Or more accurately, they believe they can follow the NetZero or Juno
  model (Free in exchange for ads being pushed to you).
 
  -forrest
 
  C F wrote:
   They believe they have advertisement revenues.
  
   On Sat, Feb 7, 2009 at 5:45 PM, Ignacio Ortega A. wrote:
  
   How Magic Jack can only charge $20 per year?
  
   do they have a call limit?
   do they have a call duration limit or limit of minutes per day?,
  
  
   Thanks
  
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-- 
http://www.linkedin.com/in/kinjaldixit

open networker
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