Re: [asterisk-users] Play a number of files to a caller

[Julian Lyndon-Smith]

Hi Tilghman , thanks for the help.

ControlPlayback can't be used with ExternalIVR, can it ?

We use ControlPlayback in our current dialplan, what I am wanting (in
concept) is to have a meetme/conference room where one of the parties
is a caller, and the other party a file to be controlplaybacked by the
caller ;)

Best I have come up with so far is to start an attendant menu, get
some curl data,  if blank WaitExten() loop back to start, if not
ControlPlayback the data. If I also background() some music file, will
that play while the loop is running ? I suspect that it will start
again from the beginning.

Julian

On 29 August 2010 18:17, Tilghman Lesher  wrote:
> On Sunday 29 August 2010 03:32:07 Julian Lyndon-Smith wrote:
>> Still can't figure out how to fastforward / rewind the current file
>> being played.
>
> core show application ControlPlayback
>
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[asterisk-users] SIP Debug Messages

[Positively Optimistic]

Is it possible to send sip messages (debug) to a file or syslog server
without having them present in the console?   If so, does anyone know what
kind of performance hit this would create.

Instance has approx 800 sip peers.
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[asterisk-users] How to Billing for MeetMe Conference?

[Zhang Shukun]

hi,Dear all
as you know, MeetMe has cdr for each attendant. but the fee is
always paid by the moderator. not by each one.
and the the members in the conference are dynamic changed.

in this scenario, how to billing for MeetMe conference? and i want to
hungup all the calls when the account fee of the moderator is not
enough.

thanks for your help!

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Re: [asterisk-users] Could MeetMe invite someone to the conference?

[Zhang Shukun]

Thank you !  Do you konw how to realtime billing for MeetMe conference?

2010/8/30 Paul Belanger :
> On Sun, Aug 29, 2010 at 10:52 PM, Zhang Shukun  wrote:
>> but i want to know if i can invite some one to the conference when i
>> already in the conference?
>>
> http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
>
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[asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[Nikhil Nair]

Hi,

I've recently had a fairly prolonged SIP registration attack, 18 hours in 
this case and often with 200 attempts per second, and suspect I've had a 
number of these in the past.  The main symptom I noticed previously was, 
because Asterisk was responding to each registration request it received, 
it was very quickly using up my 448 kbps upload limit for my home ADSL 
connection: any further traffic (i.e. anything I did) was then 
experiencing significant packet loss.

Anyway, I've now implemented the "7 steps to better Asterisk security" 
that I found on the Digium website (deny/permit, alwaysauthreject etc.), 
and have been looking at fail2ban.  However, when I attempted to install 
it (following the instructions I found on a page about fail2ban with 
Asterisk), I ran into a couple of issues.

FWIW, I'm using Asterisk 1.4.21.2~dfsg-3+lenny1 on Debian.

First, I tried uncommenting the line in /etc/asterisk/logger.conf, i.e.
dateformat=%F %T
and verified that the date format in /var/log/asterisk/full had, indeed, 
changed (after I did an asterisk -rx 'logger reload', of course).  It had 
changed: it now started with the year, instead of Aug; however, the 
parentheses were still there, whereas the instructions seemed to indicate 
that they'd disappear when this line was used in logger.conf.

At that point, I presumed I'd have to use syslog, after all, as that was 
given as the only alternative if the date format couldn't be fixed 
properly.  That wasn't my preference, but it was still workable.

The second snag I found was that, after I fixed sip.conf to include 
appropriate deny= and permit= lines and alwaysauthreject=yes, the failed 
registration attempts were no longer being logged in 
/var/log/asterisk/full at all, despite my having the line
full => notice,warning,error,debug,verbose
in the logfiles section of logger.conf.

It seems that the attack was coming from a region that was denied in 
sip.conf.  This is obviously no problem from the security point of view, 
as the attempt would inevitably fail; however, my issue isn't that the 
attack might succeed, but rather, that by responding to the attack at all, 
Asterisk is grinding my internet connection to a halt.  And Asterisk is, 
indeed, still responding, rather than just ignoring the attempts.

Is there a way to get Asterisk to log failed SIP registration attempts 
that come from a denied IP address?  Or a way to get it to simply ignore 
such attempts?

I have a feeling that a major Debian release has come out recently, and 
passed me by.  I'm wondering if that contains Asterisk 1.6, and, if so, 
whether all these issues (date format as well as logging sip registration 
attempts from denied IP addresses) might be present in that release.  That 
would certainly present a neat solution - just upgrade my machine!

Any input very welcome.

Oh, if it's of any interest: I worked out what was going on by using 
tshark (terminal version of wireshark).  In 20 seconds, it captured well 
over 7000 packets, rather than the 30 or so I was expecting - and these 
included about 4000 packets arriving from one host with SIP registration 
attempts, fully 200 per second...

Best,

Nikhil.


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Re: [asterisk-users] Web-meetme

[Tim Nelson]

- "Flavio Miranda"  wrote: 
> 
> Hi all, 
> 
> I am trying to set up Web-meetme in Asterisk 1.6. After some attemps, I am 
> receiving the message: DB Error: connect failed 
> What could be ? 



It's very likely the connection failed to your database...  


Check your database name and credentials. 


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Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[Gordon Henderson]

On Mon, 30 Aug 2010, Nikhil Nair wrote:

> Hi,
>
> I've recently had a fairly prolonged SIP registration attack, 18 hours in
> this case and often with 200 attempts per second, and suspect I've had a
> number of these in the past.

Almost everyone has - read the fine archives, then google for sipvicious 
because that's what they're using.

18-hours eh? This recent one broke my record - just under 3 days. See 
this:

   http://unicorn.drogon.net/hack1.png

green is inbound - all from one site belonging to a Romanian telephone/ISP 
- sustained from Thursday evening until Sunday lunchtime.

The real issue here is that most of the hackers I've had attack me and my 
clients are using an older version of sipvicious - and the problem with 
that is that it will not go away when you firewall it, so using fail2ban, 
etc. is a waste of time against it - it might protect your asterisk box, 
but it won't protect your network. In the above case I had, I added 
firewalling into the router where the blue-line output line went to zero 
on Thursday evening (I was playing with it on Friday morning which is why 
there's output from it then)

However because the ISP in this case counts all traffic coming in, it 
counted against their monthly allowance - that for me and my clients is 
going to be the killer more than anything else - we can firewall against 
these things, but sipvicious doesn't care - it just keeps on pumping the 
data towards you and your ISP keeps on incrementing the counters and 
billing you for it (and I've yet to find a UK ISP who will put in a block 
at their border against this sort of thing - actually, I know one, but 
they're too expensive for most).

At least the older versions of sipvicious behave this way, but when do 
criminals bother to upgrade their software? They don't seem to care - 
they've already stolen resources, so it's no big issue to them.

This problems is not going to go away - if anything, I reckon it will get 
worse in the near future. Fail2ban, etc. is not going to protect you from 
broken versions of sipvicious. Anyone who can not firewall their inbound 
SIP port to a known set of IP addresses is inviting attack, and they will 
be attacked. The sipvicious tools make scanning very easy indeed, so you 
will have to take additional measures if you want to save your bandwidth 
and sanity.

So.. Get a copy of the sipvicious code from http://blog.sipvicious.org/ 
(or directly from http://code.google.com/p/sipvicious/ ) and learn how to 
use svcrash.py as that's the only thing that's going to ultimately stop a 
long-term attack on your site. For now, anyway.

Gordon

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[asterisk-users] help with dialplan

[Todd Reese]

  Hi all,

I've been have problems with getting this system on line and would like 
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines 
listed as dialout1, dialout2, dialout3. This version of asterisk is 
1.6.2.11.  Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten => 
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

exten => 30,1,Macro(oneline,${GMNETPHONE0})
exten => 31,1,Macro(oneline,${GMNETPHONE1})
exten => 32,1,Macro(oneline,${GMNETPHONE2})
exten => 33,1,Macro(oneline,${GMNETPHONE3})
exten => 34,1,Macro(oneline,${GMNETPHONE4})
exten => 35,1,Macro(oneline,${GMNETPHONE5})
exten => 36,1,Macro(oneline,${GMNETPHONE6})
exten => 37,1,Macro(oneline,${GMNETPHONE7})

exten => 40,1,Macro(oneline,${QPHONE0})
exten => 41,1,Macro(oneline,${QPHONE1})
exten => 42,1,Macro(oneline,${QPHONE2})
exten => 43,1,Macro(oneline,${QPHONE3})
exten => 44,1,Macro(oneline,${QPHONE4})
exten => 45,1,Macro(oneline,${QPHONE5})
exten => 46,1,Macro(oneline,${QPHONE6})
exten => 47,1,Macro(oneline,${QPHONE7})

exten => 150,1,Macro(oneline,${EXTERNPHONE0})




[macro-oneline]
exten => s,1,Set(CHANNEL(musicclass)=default)
exten => s,n,Dial(${ARG1},20,Ttr)
exten => s,n,Voicemail(${MACRO_EXTEN})
exten => s,n,Hangup
exten => s,102,Vo

Re: [asterisk-users] SIP Debug Messages

[Paul Belanger]

On Mon, Aug 30, 2010 at 3:19 AM, Positively Optimistic
 wrote:
> Is it possible to send sip messages (debug) to a file or syslog server
> without having them present in the console?
>
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt

Specifically,

*CLI> sip set debug peer xxx

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Re: [asterisk-users] help with dialplan

[Paul Belanger]

On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese  wrote:
> I've been have problems with getting this system on line and would like
> to acquire some help with the extensions.conf.
>
Post a debug log of the problem:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
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Re: [asterisk-users] help with dialplan

[Bryant Zimmerman]

Todd

How do you have the context in the phones sip configs set?

Bryant

From: "Todd Reese" trees...@gmail.com

Hi all,

I've been have problems with getting this system on line and would like 
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines 
listed as dialout1, dialout2, dialout3. This version of asterisk is 
1.6.2.11. Below is the extensions.conf file.

[globals]

QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133

[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&$
{QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup

[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${
ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten => 
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNET
PHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

exten => 30,1,Macro(oneline,${GMNETPHONE0})
exten => 31,1,Macro(oneline,${GMNETPHONE1})
exten => 32,1,Macro(oneline,${GMNETPHONE2})
exten => 33,1,Macro(oneline,${GMNETPHONE3})
exten => 34,1,Macro(oneline,${GMNETPHONE4})
exten => 35,1,Macro(oneline,${GMNETPHONE5})
exten => 36,1,Macro(oneline,${GMNETPHONE6})
exten => 37,1,Macro(oneline,${GMNETPHONE7})

exten => 40,1,Macro(oneline,${QPHONE0})
exten => 41,1,Macro(oneline,${QPHONE1})
exten => 42,1,Macro(oneline,${QPHONE2})
exten => 43,1,Macro(oneline,${QPHONE3})
exten => 44,1,Macro(oneline,${QPHONE4})
exten => 45,1,Macro(oneline,${QPHONE5})
exten => 46,1,Macro(oneline,${QPHONE6})
exten => 47,1,Macro(oneline,${QPHONE7})

exten => 150,1,Macro(oneline,${EXTERNPHONE0})

[macro-oneline]
exten => s,1,Set(CHANNEL(musicclass)=default)
exten => 

Re: [asterisk-users] help with dialplan

[Todd Reese]

 Here is the sip.conf portion for extension 150

[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device <150>
accountcode=
call-limit=50


On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:

Todd

How do you have the context in the phones sip configs set?

Bryant
*
From*: "Todd Reese" trees...@gmail.com 

Hi all,

I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =>
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

exten => 30,1,Macro(oneline,${GMNETPHONE0})
exten => 31,1,Macro(oneline,${GMNETPHONE1})
exten => 32,1,Macro(oneline,${GMNETPHONE2})
exten => 33,1,Macro(oneline,${GMNETPHONE3})
exten => 34,1,Macro(oneline,${GMNETPHONE4})
exten => 35,1,Macro(oneline,${GMNETPHONE5})
exten => 36,1,Macro(oneline,${GMNETPHONE6})
exten => 37,1,Macro(oneline,${GMNETPHONE7})

exten => 40,1,Macro(oneline,${QPHONE0})
exten => 

Re: [asterisk-users] Play a number of files to a caller

[Elliot Otchet]

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian 
Lyndon-Smith
Sent: Monday, August 30, 2010 2:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play a number of files to a caller

Hi Tilghman , thanks for the help.

ControlPlayback can't be used with ExternalIVR, can it ?

We use ControlPlayback in our current dialplan, what I am wanting (in
concept) is to have a meetme/conference room where one of the parties is a 
caller, and the other party a file to be controlplaybacked by the caller ;)

Best I have come up with so far is to start an attendant menu, get some curl 
data,  if blank WaitExten() loop back to start, if not ControlPlayback the 
data. If I also background() some music file, will that play while the loop is 
running ? I suspect that it will start again from the beginning.

Julian

On 29 August 2010 18:17, Tilghman Lesher  wrote:
> On Sunday 29 August 2010 03:32:07 Julian Lyndon-Smith wrote:
>> Still can't figure out how to fastforward / rewind the current file
>> being played.
>
> core show application ControlPlayback
>
Using the AGI approach, you can also control the start and stop of Music on 
Hold.  Using this approach, I was able to mix Music on Hold and a program I 
wrote to check for new mail (my event in this case was an IMAP NewMail notify 
event) - when new mail was identified, I was able to convert it using a TTS 
engine and play it back for the user using ControlPlayback.  The start and stop 
of Music on Hold was key for me.  It was non-blocking audio playback.

YMMV.

Regards,

Elliot

This message is intended only for the use of the individual (s) or entity to 
which it is addressed and may contain information that is privileged, 
confidential, and/or proprietary to Calling Circles LLC and its affiliates. If 
the reader of this message is not the intended recipient, you are hereby 
notified that any dissemination, distribution, forwarding or copying of this 
communication is prohibited without the express permission of the sender. If 
you have received this communication in error, please notify the sender 
immediately and delete the original message.

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[asterisk-users] Maximum Wait Time queue option

[Tino]

Hello,

Is there any option to set the maximum number of seconds a caller can wait
in a queue before being pulled out ?

Thanks
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Re: [asterisk-users] Play a number of files to a caller

[Kevin P. Fleming]

On 08/30/2010 01:58 AM, Julian Lyndon-Smith wrote:

> ControlPlayback can't be used with ExternalIVR, can it ?
> 
> We use ControlPlayback in our current dialplan, what I am wanting (in
> concept) is to have a meetme/conference room where one of the parties
> is a caller, and the other party a file to be controlplaybacked by the
> caller ;)
> 
> Best I have come up with so far is to start an attendant menu, get
> some curl data,  if blank WaitExten() loop back to start, if not
> ControlPlayback the data. If I also background() some music file, will
> that play while the loop is running ? I suspect that it will start
> again from the beginning.

No, ControlPlayback can't be used with ExternalIVR, because ExternalIVR
passes *all* DTMF input from the connected channel to the ExternalIVR
process, since that is necessary to be able to interrupt playback.

It wouldn't be terribly hard to extend the ExternalIVR protocol to allow
the external process to rewind/fast-forward the file being played back,
and then it could do that based on receiving DTMF input.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] help with dialplan

[Todd Reese]

  Here's a debug for extension 150



[Aug 30 11:34:53] VERBOSE[2099] config.c:   == Parsing 
'/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c: 
Parsing /etc/asterisk/logger.conf
[Aug 30 11:34:53] VERBOSE[2099] config.c:   == Found
[Aug 30 11:34:53] VERBOSE[2099] logger.c:  Asterisk Event Logger restarted
[Aug 30 11:34:53] VERBOSE[2099] logger.c:  Asterisk Queue Logger restarted
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  0 [ 38]: OPTIONS 
sip:76.122.117.31:5060 SIP/2.0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  1 [ 44]: Via: 
SIP/2.0/UDP 64.34.245.174:5060;branch=0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  2 [ 38]: From: 
sip:pin...@voip.com;tag=7c9c6206
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  3 [ 26]: To: 
sip:76.122.117.31:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  4 [ 47]: Call-ID: 
89c833e4-9c8f2516-5bb...@64.34.245.174
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  5 [ 15]: CSeq: 1 OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  6 [ 17]: 
Content-Length: 0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  7 [  0]:
[Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with 
address 10.0.1.102:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 
89c833e4-9c8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP)
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Received OPTIONS (3) - 
Command in SIP OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' 
onto UDP socket destined for 64.34.245.174:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be 
handled, bad request: 89c833e4-9c8f2516-5bb...@64.34.245.174
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  0 [ 38]: OPTIONS 
sip:76.122.117.31:5060 SIP/2.0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  1 [ 44]: Via: 
SIP/2.0/UDP 64.34.245.174:5060;branch=0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  2 [ 38]: From: 
sip:pin...@voip.com;tag=1f9c6206
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  3 [ 26]: To: 
sip:76.122.117.31:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  4 [ 47]: Call-ID: 
89c833e4-3f8f2516-5bb...@64.34.245.174
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  5 [ 15]: CSeq: 1 OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  6 [ 17]: 
Content-Length: 0
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Header  7 [  0]:
[Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with 
address 10.0.1.102:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 
89c833e4-3f8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP)
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c:  Received OPTIONS (3) - 
Command in SIP OPTIONS
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' 
onto UDP socket destined for 64.34.245.174:5060
[Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be 
handled, bad request: 89c833e4-3f8f2516-5bb...@64.34.245.174
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog 
'89c833e4-806e2516-79b...@64.34.245.174'
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 
89c833e4-806e2516-79b...@64.34.245.174
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog 
'89c833e4-236e2516-79b...@64.34.245.174'
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 
89c833e4-236e2516-79b...@64.34.245.174
[Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
<--- SIP read from UDP:97.80.176.231:5060 --->



<->
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  0 [  0]:
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:Body  0 [  0]:
[Aug 30 11:34:54] VERBOSE[2079] chan_sip.c:
<--- SIP read from UDP:97.80.176.231:5060 --->
INVITE sip:6789542...@qci.homeip.net SIP/2.0
Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
From: "ATAP" ;tag=ee0cedf5f71d40f9
To: 
Contact: 
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 62f35b2ee0ada...@10.11.17.24
CSeq: 21395 INVITE
User-Agent: Grandstream GXP2000 1.2.3.5
Max-Forwards: 70
Allow: 
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 345

v=0
o=150 8000 8000 IN IP4 10.11.17.24
s=SIP Call
c=IN IP4 10.11.17.24
t=0 0
m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20

<->
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  0 [ 44]: INVITE 
sip:6789542...@qci.homeip.net SIP/2.0
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  1 [ 64]: Via: 
SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
[Aug 30 11:34:54] DEBUG[2079] chan_sip.c:  Header  2 [ 58]: From: "ATAP" 
;tag=ee0

Re: [asterisk-users] help with dialplan

[Bryant Zimmerman]

Todd

Your context must be set to where you want your extension to start each 
time it dials out. Without getting into your dialplan code too much try 
changing the context to point to dialout1

context=dialout1

If dialout1 is working you should be able to dial.

The best way to handle this is to create a context that when you dial from 
your phones it decieds if you have dialed an extension or an external 
number and then routes the call correclty. This way you can pickup an 
extension and dial either and get the desired results.

Bryant


 From: "Todd Reese" 
Sent: Monday, August 30, 2010 11:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] help with dialplan

Here is the sip.conf portion for extension 150

[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device <150>
accountcode=
call-limit=50

On 8/30/2010 10:37 AM, Bryant Zimmerman wrote: Todd

How do you have the context in the phones sip configs set?

Bryant

From: "Todd Reese" trees...@gmail.com

Hi all,

I've been have problems with getting this system on line and would like 
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines 
listed as dialout1, dialout2, dialout3. This version of asterisk is 
1.6.2.11. Below is the extensions.conf file.

[globals]

QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133

[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&$
{QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup

[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => 
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${
ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten => 
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNET
PHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0

Re: [asterisk-users] TELUS British Columbia PRI Settings

[Jeremy.Hellstrom]

It is a half turned up PRI, so 1-12 should be correct?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: August 28, 2010 12:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings

On Sat, Aug 28, 2010 at 01:32:13PM -0400, Andres wrote:
> On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
> >
> > I’m not surprised both the conf file and myself are confused.
> >
> > __
> >
> > I still end up with messages telling me that a dchannel cannot be 
> > found. Any other suggestions?
> >
> > Thanks, Jeremy
> >
> >
> I suggest you start over as I can see you are still confused about the 
> content of the 2 files and still trying to use spanmap.
> 
> Start with /etc/dahdi/system.conf:
> -
> loadzone=us
> defaultzone=us
> span=1,1,0,esf,b8zs
> bchan=1-23
> hardhdlc=24
> 

Basically fine (hardhdlc, indeed?)

> 
> ...and /etc/asterisk/chan_dahdi.conf:

Almost OK. Just one thing missing:

> 

[channels]

> language=en
> context=from-pstn
> switchtype=national
> signalling = pri_cpe
> group=1
> channel => 1-12
> ---

(1-12? not 1-23?)

-- 
   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] TELUS British Columbia PRI Settings

[Jeremy.Hellstrom]

I tried those as you said, deleting my failed attempt.  I've found that
using hardhdlc=24 generates an error and reminds me that FXO uses FXS
signalling and vice versa when running dadhi_restart, which seems to
indicate that it is the wrong variable name.

I also notice that if I change that variable to dchan in system.conf, I
receive no error but no matter what value I put in Asterisk looks at
channel 24 for the dchannel and as this is a half turned up PRI I have
suspicions that the dchan is not 24 though I still need TELUS to get
back to me to confirm that. 

Any other ideas as to what I am doing wrong or if I am making a bad
assumption?

Thanks, Jeremy

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: August 28, 2010 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings

On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
>
> I'm not surprised both the conf file and myself are confused.
>
> __
>
> I still end up with messages telling me that a dchannel cannot be 
> found. Any other suggestions?
>
> Thanks, Jeremy
>
>
I suggest you start over as I can see you are still confused about the 
content of the 2 files and still trying to use spanmap.

Start with /etc/dahdi/system.conf:
-
loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
hardhdlc=24


...and /etc/asterisk/chan_dahdi.conf:

language=en
context=from-pstn
switchtype=national
signalling = pri_cpe
group=1
channel => 1-12
---

That is the most basic stuff you need to get the PRI up.

Andres
http://www.neuroredes.com

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Re: [asterisk-users] Maximum Wait Time queue option

[Warren Selby]

On Mon, Aug 30, 2010 at 10:31 AM, Tino  wrote:

> Hello,
>
> Is there any option to set the maximum number of seconds a caller can wait
> in a queue before being pulled out ?
>
> Thanks
>
>

In the Queue() command itself there is a timeout parameter.  From your
asterisk box, try running:

asterisk -rx "core show application Queue"

and pay attention to the timeout parameter.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Maximum Wait Time queue option

[Tino]

Thanks Warren for your help

On Mon, Aug 30, 2010 at 9:21 PM, Warren Selby  wrote:

> On Mon, Aug 30, 2010 at 10:31 AM, Tino  wrote:
>
>> Hello,
>>
>> Is there any option to set the maximum number of seconds a caller can wait
>> in a queue before being pulled out ?
>>
>> Thanks
>>
>>
>
> In the Queue() command itself there is a timeout parameter.  From your
> asterisk box, try running:
>
> asterisk -rx "core show application Queue"
>
> and pay attention to the timeout parameter.
>
> --
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[J. Oquendo]

Gordon Henderson wrote:

> >
> > So.. Get a copy of the sipvicious code from http://blog.sipvicious.org/ 
> > (or directly from http://code.google.com/p/sipvicious/ ) and learn how to 
> > use svcrash.py as that's the only thing that's going to ultimately stop a 
> > long-term attack on your site. For now, anyway.
> >
> > Gordon
> >   
>   
You're wrong when you state: "that's the only thing that's going to
ultimately stop" The fact of the matter is, its quite simple to block
attackers without relying on anything other than good old fashioned
systems/network administration.

>From the onset, if possible a "block all" "allow in whom_I_specify"
should be the Golden Rule on any environment however, in the real world
there are times when we can't just do something as simple as that. So
what's the next best thing? Good old fashioned administration:

# tail -n 10 /var/log/asterisk/messages
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"yzlj"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zdcu"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zdur"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zmug"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zoej"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zpcp"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zxnj"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zygq"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zyjb"' failed for '69.72.242.170' - No
matching peer found
[Aug 29 19:13:36] NOTICE[21056] chan_sip.c: Registration from
'"zynh"' failed for '69.72.242.170' - No
matching peer found

How about a little cron script without having to install anything? You
could run it off the hour:

rightnow=`date "+%Y-%m-%d %k"`

grep $rightnow /var/log/asterisk/messages |\
awk '/No matching peer/' | sed's:'\''::g' |\
uniq | awk '{print "iptables -A INPUT -s "$1" -j DROP"}'| sh

I've done my own IPS/IDS and honeypots on Asterisk and I can tell you
there are other ways to minimize the attempts and the attacks without
even running ANYTHING against your machine. I can tell you from
EXPERIENCE and watching and analyze about 2-3 years worth of VoIP
attacks, you'd be extremely wrong to think that sipvicious is the only
tool in someone's arsenal. Secondly, I've seen patient attackers test
accounts 1 at a time so don't think for a moment that by solely running
sipvicious and checking the results, you're in the clear.


| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-08 | 09:15:03   |
2010-08-08 | 09:15:03  | 125.71.212.123  | 1|
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-23 | 01:28:45   |
2010-08-23 | 01:28:45  | 82.201.218.31   | 1|


mysql> use arkeos

Database changed
mysql> select * from bruteforcers where start_date like '%2010-08%';
+--++++---+-+--+
| hostid   | start_date | start_time |
stop_date  | stop_time | attacker| attempts |
+--++++---+-+--+
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-02 | 12:28:22   |
2010-08-02 | 12:58:27  | 88.42.207.98| 54644|
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-04 | 11:46:29   |
2010-08-04 | 11:48:18  | 93.35.113.170   | 9975 |
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-04 | 13:08:48   |
2010-08-04 | 13:09:16  | 210.22.14.113   | 4187 |
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-06 | 01:51:15   |
2010-08-06 | 02:26:43  | 187.63.73.3 | 142904   |
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-08 | 09:15:03   |
2010-08-08 | 09:15:03  | 125.71.212.123  | 1|
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-08 | 15:42:59   |
2010-08-08 | 17:07:54  | 217.174.169.29  | 108120   |
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-08 | 18:20:40   |
2010-08-08 | 18:53:58  | 61.218.212.75   | 79195|
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-08 | 19:07:25   |
2010-08-08 | 19:39:52  | 72.166.143.8| 50073|
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-10 | 19:20:27   |
2010-08-10 | 19:21:02  | 61.164.41.144   | 2797 |
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-11 | 10:54:14   |
2010-08-11 | 12:24:36  | 222.73.93.143   | 128352   |
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-11 | 16:07

Re: [asterisk-users] help with dialplan

[Todd Reese]

 Unfortunately, that didn't work.  The phone is still giving me a 404 
error.


I have my own system that is 1.6.2.7 with Grandstream phones that works 
fine.  Using it as a guide, I built this server for a client which also 
has Grandstream phones.


Last week, it dialed out fine.  Since the weekend, no dialing at all.

On 8/30/2010 11:42 AM, Bryant Zimmerman wrote:

Todd

Your context must be set to where you want your extension to start 
each time it dials out. Without getting into your dialplan code too 
much try changing the context to point to dialout1


context=dialout1

If dialout1 is working you should be able to dial.

The best way to handle this is to create a context that when you dial 
from your phones it decieds if you have dialed an extension or an 
external number and then routes the call correclty. This way you can 
pickup an extension and dial either and get the desired results.


Bryant




*From*: "Todd Reese" 
*Sent*: Monday, August 30, 2010 11:20 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] help with dialplan

Here is the sip.conf portion for extension 150

[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device <150>
accountcode=
call-limit=50


On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:

Todd

How do you have the context in the phones sip configs set?

Bryant
*
From*: "Todd Reese" trees...@gmail.com 

Hi all,

I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>
s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =>
s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout

[asterisk-users] Wifi + SIP + Asterisk

[Narendra Sisodiya]

In my old office, for conference purpose , gotomeeting was used. also for
the lecture delivery, same gobomeeting was used, most the time , we need to
listen voice only. also, we use to share desktop screen.
But as far as I know SIP is the standard for video telephony. SIP can handle
video +Audio.
now, I am thinking that, I can give solution like selling a server which has
Asterisk over it. AFAIK, Asterisk can handle VIOP calls. Now system will
load some GUI application from there he can add remove users. Now at the
same time I want to give small device which has
Wifi + LCD (for video) + Android + webcam+Sipdroid or IMSdroid. China can
make such device in less then 100 dollar. now, every customer will be given
with a unique number of video calling. So I am thinking for selling such
Office-Videotelefony solution based on opensoure and open standard. this is
one of the many idea which i am trying to explore. I am  a totally new user
to Asterisk work but working as FOSS evangelist and developer for last 2
years. I am totally aware of 'open ecosystem'.
Please share your thought on this idea and obstacle.

PS: desktop screen sharing will be possible with some hacks.. (not
impossible)
-- 
┌─┐
│Narendra Sisodiya
│http://narendrasisodiya.com
└─┘
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Re: [asterisk-users] help with dialplan

[Paul Belanger]

On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese  wrote:
>  Here's a debug for extension 150
>
In the future, simply attach your debug log to your email.  Here is
your problem:

[Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension
'6789542133' rejected because extension not found in context
'extensions.conf'.


-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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Re: [asterisk-users] help with dialplan

[Alex Bell]

possibly check you spelling:  [from-interal] -> [dialout1]
include => from-internal

??

On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese  wrote:

>  Hi all,
>
> I've been have problems with getting this system on line and would like
> to acquire some help with the extensions.conf.
>
> My current problem is that the phones won't dialout.on the VOIP lines
> listed as dialout1, dialout2, dialout3. This version of asterisk is
> 1.6.2.11.  Below is the extensions.conf file.
>
>
> [globals]
>
>
>
> QPHONE0=SIP/10
> QPHONE1=SIP/11
> QPHONE2=SIP/12
> QPHONE3=SIP/13
> QPHONE4=SIP/14
> QPHONE5=SIP/15
> QPHONE6=SIP/16
> QPHONE7=SIP/17
>
> ACAPHONE0=SIP/20
> ACAPHONE1=SIP/21
> ACAPHONE2=SIP/22
> ACAPHONE3=SIP/23
> ACAPHONE4=SIP/24
> ACAPHONE5=SIP/25
> ACAPHONE6=SIP/26
> ACAPHONE7=SIP/27
>
> GMNETPHONE0=SIP/30
> GMNETPHONE1=SIP/31
> GMNETPHONE2=SIP/32
> GMNETPHONE3=SIP/33
> GMNETPHONE4=SIP/34
> GMNETPHONE5=SIP/35
> GMNETPHONE6=SIP/36
> GMNETPHONE7=SIP/37
>
> EXTERNPHONE0=SIP/150
>
> CPHONE1=SIP/1678000
> CPHONE2=SIP/177
>
> EMERGENCY=0
> EMERGENCY_TRUNK=DAHDI/G1
> ; Change this for production use:
> EMERGENCY_NUM=6789542133
>
>
> [from-pstn]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
>
>
>
> [from-pstn1]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
> [from-pstn2]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
> [from-pstn3]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
> [from-pstn4]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming3,s,1)
>
> [from-pstn5]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming2,s,1)
>
> [from-pstn6]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
> [from-pstn7]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
> [from-pstn8]
> exten => s,1,Set(FROM_DID="678000)
> exten => s,n,NoOp(id is ${FROM_DID})
> exten => s,n,Goto(incoming1,s,1)
>
>
> [incoming1]
> include => from-internal
> include => parkedcalls
> exten => s,1,Answer
> exten => s,n,Wait(1)
> exten => s,n,Set(CHANNEL(musicclass)=QCI)
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n,Background(thank-you-for-calling)
> exten =>
>
> s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
> exten => s,n,Hangup
>
>
> [incoming2]
> include => from-internal
> include => parkedcalls
> exten => s,1,Answer
> exten => s,n,Wait(1)
> exten => s,n,Set(CHANNEL(musicclass)=QCI)
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n,Background(thank-you-for-calling)
> exten =>
>
> s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
> exten => s,n,Hangup
>
> [incoming3]
> include => from-internal
> include => parkedcalls
> exten => s,1,Answer
> exten => s,n,Wait(1)
> exten => s,n,Set(CHANNEL(musicclass)=QCI)
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n,Background(thank-you-for-calling)
> exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
> exten =>
>
> s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
> exten => s,n,Hangup
>
> [from-interal]
> include => dialout1
> include => dialout2
> include => dialout3
> include => parkedcalls
> include => intercom
>
> exten => 10,1,Macro(oneline,${QPHONE0})
> exten => 11,1,Macro(oneline,${QPHONE1})
> exten => 12,1,Macro(oneline,${QPHONE2})
> exten => 13,1,Macro(oneline,${QPHONE3})
> exten => 14,1,Macro(oneline,${QPHONE4})
> exten => 15,1,Macro(oneline,${QPHONE5})
> exten => 16,1,Macro(oneline,${QPHONE6})
> exten => 17,1,Macro(oneline,${QPHONE7})
>
> exten => 20,1,Macro(oneline,${ACAPHONE0})
> exten => 21,1,Macro(oneline,${ACAPHONE1})
> exten => 22,1,Macro(oneline,${ACAPHONE2})
> exten => 23,1,Macro(oneline,${ACAPHONE3})
> exten => 24,1,Macro(oneline,${ACAPHONE4})
> exten => 25,1,Macro(oneline,${ACAPHONE5})
> exten => 26,1,Macro(oneline,${ACAPHONE6})
> exten => 27,1,Macro(oneline,${ACAPHONE7})
>
> exten => 30,1,Macro(oneline,${GMNETPHONE0})
> exten => 31,1,Macro(oneline,${GMNETPHONE1})
> exten => 32,1,Macro(oneline,${GMNETPHONE2})
> exten => 33,1,Macro(oneline,${GMNETPHONE3})
> exten => 34,1,Macro(oneline,${GMNETPHONE4})
> exten => 35,1,Macro(oneline,${GMNETPHONE5})
> exten => 36,1,Macro(oneline,${GMNETPHONE6})
> exten => 37,1,Macro(oneline,${GMNETPHONE7})
>
> exten => 40,1,Macro(oneline,${QPHONE0})
> exten => 41,1,Macro(oneline,${QPHONE1})

Re: [asterisk-users] help with dialplan

[Todd Reese]

  I actually found that one and corrected it.  I have replaced the 
context with the from-internal, remote, and dialout1.  Each has produced 
the same results of a 404 error.




On 8/30/2010 2:10 PM, Paul Belanger wrote:
> On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese  wrote:
>>   Here's a debug for extension 150
>>
> In the future, simply attach your debug log to your email.  Here is
> your problem:
>
> [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension
> '6789542133' rejected because extension not found in context
> 'extensions.conf'.
>
>


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Re: [asterisk-users] TELUS British Columbia PRI Settings

[Andres]

On 8/30/2010 11:42 AM, jeremy.hellst...@synovate.com wrote:
> I tried those as you said, deleting my failed attempt.  I've found that
> using hardhdlc=24 generates an error and reminds me that FXO uses FXS
> signalling and vice versa when running dadhi_restart, which seems to
> indicate that it is the wrong variable name.
>
That makes no sense.  You are configuring a PRI, not an analog line.  
You should not be getting any messages regarding FXO or FXS.  Take a 
look at the file again and delete any reference to analog channels if 
you are not using them.

Andres
http://www.neuroredes.com
> I also notice that if I change that variable to dchan in system.conf, I
> receive no error but no matter what value I put in Asterisk looks at
> channel 24 for the dchannel and as this is a half turned up PRI I have
> suspicions that the dchan is not 24 though I still need TELUS to get
> back to me to confirm that.
>
> Any other ideas as to what I am doing wrong or if I am making a bad
> assumption?
>
> Thanks, Jeremy
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
> Sent: August 28, 2010 10:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings
>
> On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
>
>> I'm not surprised both the conf file and myself are confused.
>>
>> __
>>
>> I still end up with messages telling me that a dchannel cannot be
>> found. Any other suggestions?
>>
>> Thanks, Jeremy
>>
>>
>>  
> I suggest you start over as I can see you are still confused about the
> content of the 2 files and still trying to use spanmap.
>
> Start with /etc/dahdi/system.conf:
> -
> loadzone=us
> defaultzone=us
> span=1,1,0,esf,b8zs
> bchan=1-23
> hardhdlc=24
> 
>
> ...and /etc/asterisk/chan_dahdi.conf:
> 
> language=en
> context=from-pstn
> switchtype=national
> signalling = pri_cpe
> group=1
> channel =>  1-12
> ---
>
> That is the most basic stuff you need to get the PRI up.
>
> Andres
> http://www.neuroredes.com
>
>


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Re: [asterisk-users] help with dialplan

[Todd Reese]

 Thanks for pointing out the misspelling.  I've corrected that and 
still no luck.


On 8/30/2010 2:33 PM, Alex Bell wrote:

possibly check you spelling:  [from-interal] -> [dialout1]
include => from-internal

??

On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese > wrote:


 Hi all,

I've been have problems with getting this system on line and would
like
to acquire some help with the extensions.conf.

My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11.  Below is the extensions.conf file.


[globals]



QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17

ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27

GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37

EXTERNPHONE0=SIP/150

CPHONE1=SIP/1678000
CPHONE2=SIP/177

EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133


[from-pstn]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)




[from-pstn1]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn2]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn3]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn4]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming3,s,1)

[from-pstn5]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming2,s,1)

[from-pstn6]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn7]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)

[from-pstn8]
exten => s,1,Set(FROM_DID="678000)
exten => s,n,NoOp(id is ${FROM_DID})
exten => s,n,Goto(incoming1,s,1)


[incoming1]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>

s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&${QPHONE6}&${QPHONE7},40,Ttr)
exten => s,n,Hangup


[incoming2]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten =>

s,n,Dial(${ACAPHONE0}&${ACAPHONE1}&${ACAPHONE2}&${ACAPHONE3}&${ACAPHONE4}&${ACAPHONE5}&${ACAPHONE6}&${ACAPHONE7},40,TTr)
exten => s,n,Hangup

[incoming3]
include => from-internal
include => parkedcalls
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(CHANNEL(musicclass)=QCI)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =>

s,n,Dial(${GMNETPHONE0}&${GMNETPHONE1}&${GMNETPHONE2}&${GMNETPHONE3}&${GMNETPHONE4}&${GMNETPHONE5}&${GMNETPHONE6}&${GMNETPHONE7},40,Ttr)
exten => s,n,Hangup

[from-interal]
include => dialout1
include => dialout2
include => dialout3
include => parkedcalls
include => intercom

exten => 10,1,Macro(oneline,${QPHONE0})
exten => 11,1,Macro(oneline,${QPHONE1})
exten => 12,1,Macro(oneline,${QPHONE2})
exten => 13,1,Macro(oneline,${QPHONE3})
exten => 14,1,Macro(oneline,${QPHONE4})
exten => 15,1,Macro(oneline,${QPHONE5})
exten => 16,1,Macro(oneline,${QPHONE6})
exten => 17,1,Macro(oneline,${QPHONE7})

exten => 20,1,Macro(oneline,${ACAPHONE0})
exten => 21,1,Macro(oneline,${ACAPHONE1})
exten => 22,1,Macro(oneline,${ACAPHONE2})
exten => 23,1,Macro(oneline,${ACAPHONE3})
exten => 24,1,Macro(oneline,${ACAPHONE4})
exten => 25,1,Macro(oneline,${ACAPHONE5})
exten => 26,1,Macro(oneline,${ACAPHONE6})
exten => 27,1,Macro(oneline,${ACAPHONE7})

exten => 30,1,Macro(oneline,${GMNETPHONE0})
exten => 31,1,M

Re: [asterisk-users] help with dialplan

[Paul Belanger]

On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese  wrote:
> Thanks for pointing out the misspelling.  I've corrected that and still no
> luck.
>
Create a new debug log with your recent changes, re-attach it the list.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] TELUS British Columbia PRI Settings

[Jeremy.Hellstrom]

The specific error message is as follows.
_
Changing signalling on channel 24 from Unused to Hardware assisted
D-channel
DAHDI_CHANCONFIG failed on channel 24: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?
_

I am using the simplified system.conf and chan_dahdi that you specified
before.

Thanks for continuing to help me and my gift for producing bizarre
errors.
Jeremy

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: August 30, 2010 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings

On 8/30/2010 11:42 AM, jeremy.hellst...@synovate.com wrote:
> I tried those as you said, deleting my failed attempt.  I've found
that
> using hardhdlc=24 generates an error and reminds me that FXO uses FXS
> signalling and vice versa when running dadhi_restart, which seems to
> indicate that it is the wrong variable name.
>
That makes no sense.  You are configuring a PRI, not an analog line.  
You should not be getting any messages regarding FXO or FXS.  Take a 
look at the file again and delete any reference to analog channels if 
you are not using them.

Andres
http://www.neuroredes.com
> I also notice that if I change that variable to dchan in system.conf,
I
> receive no error but no matter what value I put in Asterisk looks at
> channel 24 for the dchannel and as this is a half turned up PRI I have
> suspicions that the dchan is not 24 though I still need TELUS to get
> back to me to confirm that.
>
> Any other ideas as to what I am doing wrong or if I am making a bad
> assumption?
>
> Thanks, Jeremy
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
> Sent: August 28, 2010 10:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings
>
> On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
>
>> I'm not surprised both the conf file and myself are confused.
>>
>> __
>>
>> I still end up with messages telling me that a dchannel cannot be
>> found. Any other suggestions?
>>
>> Thanks, Jeremy
>>
>>
>>  
> I suggest you start over as I can see you are still confused about the
> content of the 2 files and still trying to use spanmap.
>
> Start with /etc/dahdi/system.conf:
> -
> loadzone=us
> defaultzone=us
> span=1,1,0,esf,b8zs
> bchan=1-23
> hardhdlc=24
> 
>
> ...and /etc/asterisk/chan_dahdi.conf:
> 
> language=en
> context=from-pstn
> switchtype=national
> signalling = pri_cpe
> group=1
> channel =>  1-12
> ---
>
> That is the most basic stuff you need to get the PRI up.
>
> Andres
> http://www.neuroredes.com
>
>


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Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[Gordon Henderson]

On Mon, 30 Aug 2010, J. Oquendo wrote:

> How about a little cron script without having to install anything? You
> could run it off the hour:
>
> rightnow=`date "+%Y-%m-%d %k"`
>
> grep $rightnow /var/log/asterisk/messages |\
> awk '/No matching peer/' | sed's:'\''::g' |\
> uniq | awk '{print "iptables -A INPUT -s "$1" -j DROP"}'| sh

Your script is fine, but I think you are missing the point I made which is 
that early versions of sipvicious are badly broken in that they will 
continue to send their attacks for days after you firewall them out.

I also posted a very effective iptables script some weeks ago if you care 
to search the archives. It works and is extremely effective in blocking 
these types of attacks - however, it will not stop a broken sipvicious 
from continuing to send data to your server, and that's the issue I have 
at present.

Any attacking site is trivial to firewall out once you've detected it, but 
firewalling it out is not going to protect you or your customers from the 
incoming data if you have to pay for it, or have a capped Internet 
connection, and the attacking site doesn't give up, even when it's 
firewalled.

A typical setup I might use for a customer in the UK is to set them up 
with an ADSL service with a 15GB/month cap. That's OK for a small office, 
or a bigger one with a line dedicated to VoIP. The attack I posted about 
earlier used up nearly 15GB of data in 3 days. Fortunately most of it was 
off-peak/weekend when it's unmetered. I arranged for their router to drop 
the packets, but the ISP still counted them.

Try it for yourself - get an early copy of SV, point it at one of your 
servers, then firewall the server against your attacking machine. 
svcrack.py will not give up.

What your script needs to do is not only get the IP address, but also the 
calling port, then launch an svcrash against that site...

... and hope that the hackers aren't getting clever and putting svcrack in 
a script that automatically re-starts it... (which I think some of them 
are now)

Gordon

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Re: [asterisk-users] TELUS British Columbia PRI Settings

[Andres]

On 8/30/2010 2:59 PM, jeremy.hellst...@synovate.com wrote:
> The specific error message is as follows.
> _
> Changing signalling on channel 24 from Unused to Hardware assisted
> D-channel
> DAHDI_CHANCONFIG failed on channel 24: Invalid argument (22)
> Did you forget that FXS interfaces are configured with FXO signalling
> and that FXO interfaces use FXS signalling?
> _
>
> I am using the simplified system.conf and chan_dahdi that you specified
> before.
>
> Thanks for continuing to help me and my gift for producing bizarre
> errors.
> Jeremy
>
from system.conf:
---
# dchan::
#   The DAHDI driver performs HDLC encoding and decoding on the
#   bundle and also performs incoming and outgoing FCS insertion
#   and verification.  'fcshdlc' is an alias for this.
# hardhdlc::
#   The hardware driver performs HDLC encoding and decoding on the
#   bundle and also performs incoming and outgoing FCS insertion
#   and verification.  Is subject to limitations and support of underlying
#   hardware.


If you are getting that error then you probably have old hardware that 
does not support 'hardhdlc', use 'dchan' instead.

Andres
http://www.neuroredes.com


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
> Sent: August 30, 2010 11:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings
>
> On 8/30/2010 11:42 AM, jeremy.hellst...@synovate.com wrote:
>
>> I tried those as you said, deleting my failed attempt.  I've found
>>  
> that
>
>> using hardhdlc=24 generates an error and reminds me that FXO uses FXS
>> signalling and vice versa when running dadhi_restart, which seems to
>> indicate that it is the wrong variable name.
>>
>>  
> That makes no sense.  You are configuring a PRI, not an analog line.
> You should not be getting any messages regarding FXO or FXS.  Take a
> look at the file again and delete any reference to analog channels if
> you are not using them.
>
> Andres
> http://www.neuroredes.com
>
>> I also notice that if I change that variable to dchan in system.conf,
>>  
> I
>
>> receive no error but no matter what value I put in Asterisk looks at
>> channel 24 for the dchannel and as this is a half turned up PRI I have
>> suspicions that the dchan is not 24 though I still need TELUS to get
>> back to me to confirm that.
>>
>> Any other ideas as to what I am doing wrong or if I am making a bad
>> assumption?
>>
>> Thanks, Jeremy
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
>> Sent: August 28, 2010 10:32 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] TELUS British Columbia PRI Settings
>>
>> On 8/28/2010 12:59 PM, jeremy.hellst...@synovate.com wrote:
>>
>>  
>>> I'm not surprised both the conf file and myself are confused.
>>>
>>> __
>>>
>>> I still end up with messages telling me that a dchannel cannot be
>>> found. Any other suggestions?
>>>
>>> Thanks, Jeremy
>>>
>>>
>>>
>>>
>> I suggest you start over as I can see you are still confused about the
>> content of the 2 files and still trying to use spanmap.
>>
>> Start with /etc/dahdi/system.conf:
>> -
>> loadzone=us
>> defaultzone=us
>> span=1,1,0,esf,b8zs
>> bchan=1-23
>> hardhdlc=24
>> 
>>
>> ...and /etc/asterisk/chan_dahdi.conf:
>> 
>> language=en
>> context=from-pstn
>> switchtype=national
>> signalling = pri_cpe
>> group=1
>> channel =>   1-12
>> ---
>>
>> That is the most basic stuff you need to get the PRI up.
>>
>> Andres
>> http://www.neuroredes.com
>>
>>
>>  
>
>


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[asterisk-users] Voicemail prompts fuzzy and quiet

[Peder]

Strange issue that I can't figure out and I am hoping someone may have some
ideas.  Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it
runs like a top and I am not going to mess with it).  *B rsyncs config from
*A.  *A dies.  I bring up *B and it all works fine, except for one issue.
Calls to voicemail are garbled and low.  Phone to phone and phone to gateway
work perfect.  If I get voicemail emailed to me, it sounds perfect.  If I
call into voicemail, the prompts and the message are garbled.  There is no
packet loss or any issues like that as all other calls sounds fine.  The
only issue is calls to voicemail from what I can tell.  We have thousands of
calls a day, so I am quite sure I would have heard if there were other
issues.  Any ideas?  My first guess would be timing, but it isn't stuttery
or anything, it is just kind of fuzzy and quiet and I can't imagine timing
would affect that.

Peder



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Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[J. Oquendo]

Gordon Henderson wrote:
> On Mon, 30 Aug 2010, J. Oquendo wrote:
>
>
>   
> I also posted a very effective iptables script some weeks ago if you care 
> to search the archives. It works and is extremely effective in blocking 
> these types of attacks - however, it will not stop a broken sipvicious 
> from continuing to send data to your server, and that's the issue I have 
> at present.
>   

Alright, so I'm slightly confused maybe I'm reading this wrong...

Someone using an older version of sipvicious was blocked and the
"blocking" of the traffic still carried a load?

If so then you should have logged into your router and simply sinkholed
him. There is nothing you can do against a flood whether or not its
sipvicious or any other program. It's the "golf ball through the water
hose" effect.

Did you try:

1) sinkholing from your router
2) Contacting your upstream to inform them of the DoS to see if they'd
sinkhole it
3) Contact the UPSTREAM of the attacking host?

+--++++---+-+--+
| hostid   | start_date | start_time |
stop_date  | stop_time | attacker| attempts |
+--++++---+-+--+
| e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-25 | 07:54:02   |
2010-08-25 | 07:55:54  | 38.99.168.133   | 16022|

8K attempts in a minute. There were times last month I'd see upwards of
40-60k per minute WHILE I played around with some of these guys in a
separate Asterisk based honeypot I created. So my confusion: "it will
not stop a broken sipvicious from continuing to send data to your
server" Even CURRENT versions of sipvicious won't stop sending data just
because you firewalled them out.

There is a pattern that many don't see unless your constantly monitoring
and watching what's going on with your logs/devices. What I see
firsthand is, there are "bruteforcers" and there are the "toll
fraudsters." Since this is a public list, I care not to discuss findings
for obvious reasons however, for those interested in that information,
feel free to send me a "non-free-mail" (meaning no Gmail, no Hotmail,
etc) message. If I get around to seeing I should share this information,
I'd gladly do so... Otherwise I won't disclose anything about honeypots,
analysis, traffic patterns, etc. Its already surprising I posted
attacker information on the forum. ;) I see all sorts of attackers,
attack vectors, numbers dialed, etc., from many of these attackers.
You'd be surprised how STUPID some are and how SMART others are.

As for your comment though, its confusing to me because if you blocked
them and they're still overwhelming you, sounds like a) you need more
bandwidth because you're on a slow connection (I'm on a DS3) or b)
server is misconfigured. On Linux tc can be your friend


-- 

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

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ruin it. If you think about that, you'll do things
differently." - Warren Buffett

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


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Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[Jian Gao]

On 10-08-30 01:53 PM, J. Oquendo wrote:
> Gordon Henderson wrote:
>
>> On Mon, 30 Aug 2010, J. Oquendo wrote:
>>
>>
>>
>> I also posted a very effective iptables script some weeks ago if you care
>> to search the archives. It works and is extremely effective in blocking
>> these types of attacks - however, it will not stop a broken sipvicious
>> from continuing to send data to your server, and that's the issue I have
>> at present.
>>
>>  
> Alright, so I'm slightly confused maybe I'm reading this wrong...
>
> Someone using an older version of sipvicious was blocked and the
> "blocking" of the traffic still carried a load?
>
> If so then you should have logged into your router and simply sinkholed
> him. There is nothing you can do against a flood whether or not its
> sipvicious or any other program. It's the "golf ball through the water
> hose" effect.
>
> Did you try:
>
> 1) sinkholing from your router
> 2) Contacting your upstream to inform them of the DoS to see if they'd
> sinkhole it
> 3) Contact the UPSTREAM of the attacking host?
>
> +--++++---+-+--+
> | hostid   | start_date | start_time |
> stop_date  | stop_time | attacker| attempts |
> +--++++---+-+--+
> | e3d8862a1f1457b8722646dbec79d0f4b7e1b2ab | 2010-08-25 | 07:54:02   |
> 2010-08-25 | 07:55:54  | 38.99.168.133   | 16022|
>
> 8K attempts in a minute. There were times last month I'd see upwards of
> 40-60k per minute WHILE I played around with some of these guys in a
> separate Asterisk based honeypot I created. So my confusion: "it will
> not stop a broken sipvicious from continuing to send data to your
> server" Even CURRENT versions of sipvicious won't stop sending data just
> because you firewalled them out.
>
> There is a pattern that many don't see unless your constantly monitoring
> and watching what's going on with your logs/devices. What I see
> firsthand is, there are "bruteforcers" and there are the "toll
> fraudsters." Since this is a public list, I care not to discuss findings
> for obvious reasons however, for those interested in that information,
> feel free to send me a "non-free-mail" (meaning no Gmail, no Hotmail,
> etc) message. If I get around to seeing I should share this information,
> I'd gladly do so... Otherwise I won't disclose anything about honeypots,
> analysis, traffic patterns, etc. Its already surprising I posted
> attacker information on the forum. ;) I see all sorts of attackers,
> attack vectors, numbers dialed, etc., from many of these attackers.
> You'd be surprised how STUPID some are and how SMART others are.
>
> As for your comment though, its confusing to me because if you blocked
> them and they're still overwhelming you, sounds like a) you need more
> bandwidth because you're on a slow connection (I'm on a DS3) or b)
> server is misconfigured. On Linux tc can be your friend
>
>
>
Joshua Stein has an great article on this topic:

http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/
-- 
Jian Gao


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[asterisk-users] Cisco 9971

[Sascha Ferley]

Hi, 

I am having a weird issue with a Cisco 9971 phone. I managed to get most of
it working, including the side car, however one of the issues is that there
seems to be some sort of side tone / beep occurring roughly every 13 seconds
or so, as if the phone is activated with call waiting.

However none of this is activated and is still making the annoying beeping
side tone. The phone does require that one runs with tcp=enable and
transport=tcp, thus turning on the presence information, which from the logs
seems to be refreshing roughly every 10 - 15 seconds. However the "Patch"
has not been compiled in, thus the information being sent to the phone is
incorrect and thus am wondering if this is what is causing this annoying
side tone.

If anyone knows, please let me know or anyone has any experience with the
9971's .. Would be awesome to get this working correctly
Thanks




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Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

[Gordon Henderson]

On Mon, 30 Aug 2010, J. Oquendo wrote:

> Gordon Henderson wrote:
>> On Mon, 30 Aug 2010, J. Oquendo wrote:
>>
>> I also posted a very effective iptables script some weeks ago if you care
>> to search the archives. It works and is extremely effective in blocking
>> these types of attacks - however, it will not stop a broken sipvicious
>> from continuing to send data to your server, and that's the issue I have
>> at present.
>
> Alright, so I'm slightly confused maybe I'm reading this wrong...
>
> Someone using an older version of sipvicious was blocked and the
> "blocking" of the traffic still carried a load?

Yes. It's UDP, they just keep on sending.

> If so then you should have logged into your router and simply sinkholed
> him. There is nothing you can do against a flood whether or not its
> sipvicious or any other program. It's the "golf ball through the water
> hose" effect.
>
> Did you try:
>
> 1) sinkholing from your router

Yes. works fine until they can send faster than the router/incoming line 
can handle the load. With a good VPS host you can trivially max-out a 
typical UK ADSL line.

> 2) Contacting your upstream to inform them of the DoS to see if they'd
> sinkhole it

Yes.

My (ADSL) upstream will not block inbound floods like this. They have a 
financial incentive not to - they get paid for the data the allow into 
their network and through to you.

I only know of one UK broadband ISP that will actively block inbound 
traffic for you and they're technically superb, but that comes with a 
price which is more than your average small business is wiling to pay. 
None of the others I know and have used will block an inbound flood of 
anything for you.

My main hosting upstream will only block such attacks when it has a 
detrimental effect on their network (and then they're very good at it) - 
last time my hosted servers got hit, they soaked up just over 30GB from a 
single VPS site in France in a 12-hour period.

> 3) Contact the UPSTREAM of the attacking host?

Yes. No reply. And in the few times I've tried, I've only ever had a reply 
from Amazon - some 18 hours after the flood started and then it took 
another 12 hours for them to stop it (well documented here in the archives 
by myself and others)

The reality is that most bulk VPS providers just don't care, or you've got 
to go through layes of their own (semi-automated) protocol to get anywhere 
(cf. Amazon)

Basically if you have to pay for inbound traffic in any shape or form 
(monthly cap, daily limit, etc.) then you're fucked when this happens.

That's why the author of Sipvicious added svcrash.py to his set of 
scripts.

Gordon

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