wow, this stuff is awesome! it's the best thing EVER! it's like the
much awaited asterisk for windows! it has fiber! fiber, people, fiber!
And it Detoxifies the body of infectious toxins! not just any
toxins, infectious toxins!
somehow i get the feeling that asterisk is going to be paying less of
with this.
-yair
On 10/22/07, Yair Hakak [EMAIL PROTECTED] wrote:
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine,
hi all,
you'll have to excuse the ignorance (i'm a software guy, not a telcom
guy..)
Is there any way to know if a channel has been answered by an automatic
system (like voicemail) rather than a human being?
Specifically, I want to use a .call to make a call on a channel and only do
something
Hello all,
Hoping someone can help me with an issue...I have i .call file which calls
out on a SIP channel and connects to an extension which dials another SIP
channel. (both via voip providers) - both to PSTN.
Problem is, hanging up the POTS phone doesn't release the channel (either
one -
hi all,
i have the following .call file:
Channel: IAX2/[EMAIL PROTECTED]/myPOTSline
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: default
Extension: 156
Priority: 1
when i drop the .call file into the
Hello all,
I'm having a bit for a problem with the dial command limit option. I have
the following dial command (executed from inside the a2billing agi)
AGI Script Executing Application: (Dial) Options: (
IAX2/[EMAIL PROTECTED]/18005551212|30|HL(6:2:0)0)
Now, from what i read in the
-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel
israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED] -- Yair Hakak
-Yair Hakak, CEOGo Telecom, Ltd., Israelisrael: (972) 54
hi list,
i need some help here...
ihave the following setup
1. openser running on port 5060 - succesfully registering endpoints. all good.
2 asterisk 1.2 running sip on port 5070 on the same machine.
3. asterisk 1.09 running sip on port 5070 a different machine.
i have 2 routes in my
/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel
israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com
/listinfo/asterisk-users-- Yair Hakak-Yair Hakak, CEOGo Telecom, Ltd., Israel
israel: (972) 54 5491266usa: (212) 202 2340[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk
in the [general] section of sip.conf
bindport=5062
well documented here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
-yair
On 6/23/06, Khaled Chehab [EMAIL PROTECTED] wrote:
How I can let asterisk listen only at port 5062 since I have ser on the same machine
run asterisk in verbose mode (-vv) and see if the digits are being properly picked up. A common problem is a DTMF type mismatch, so the keypresses may not be getting to the server.
-yair
On 5/1/06, Jim Lynch [EMAIL PROTECTED] wrote:
I've enabled voice mail for extension 200 in the extensions
hi,
SER is less about the number of callers than it is about the number of registered sip clients. Without NAT issues a pizza box server with SER can essentially register an unlimited number of SIP clients.
With larger numbers of SIP clients i find SER handles them much better than asterisk.
Now,
anyone else having issues with voipjet? i am getting nothing but dead air and a hanging iax channel.
-yair
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
to ser.
Cheick Zerbo
Corbimas.com
[EMAIL PROTECTED]
From: Yair Hakak [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comTo: Asterisk Users List
asterisk-users@lists.digium.comSubject: [Asterisk-Users] re: questions about sip requests
hi all,
I keep asking the question and getting no replies, so i'll keep asking :-)
In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically
rewritehostport(myIP:5070); (asterisk running on port 5070) asterisk picks up the request and matches it to the
Hi Jean-Michel,
have you tried upgrading? can you confirm this behavior? It seems to me this is a major issue for those of us running SER + asterisk, and who dont want to configure each SIP client in SER and asterisk separately.
-yair
On 2/5/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
In
hello all,
i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the
hi,
why cant you just playback what you want to play specifically before
going to MOH, i.e.
exten = 6000,1,Answer
exten - 6000,2, Playback()
exten = 6000,3,MusicOnHold()
sorry if i'm missing something...
-yair
On 1/24/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello All,
Does
lukeuse the wiki.
(always wanted to do that)
http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone
hope this helps,
yair
On 1/6/06, luke devon [EMAIL PROTECTED] wrote:
HI ,
I installed asterisk in fedora core 3 machine perfectly. and i have 10 units
of GrandStream IP
what is your question? you must set up extensions yourself in
extensions.conf...you can set the extensions to whatever you want
(say, if you are replacing an existing PBX and want the users to have
the same extensions).
-yair
On 12/3/05, Vladimir Montealegre [EMAIL PROTECTED] wrote:
*411
hello all,
having a little problem..
asterisk and ser on the same box, SER on 5060 and asterisk on 5070.
SER is set up to forward everything to asterisk.
in 1.07 my sip.conf looked like this:
[general]
port = 5070 ; Port to bind to
disallow=all; Disallow
look, you get what you pay for.
excellent value for the price, but i've found they need more
handholding than others (sometimes they need to be rebooted, they
freeze up, etc). i'm phasing out in favor of pap2 units and analog
phones.
i've never had a problem with audio quality, however, audio
hi all,
compiling 1.2 from CVS i get the following error in asterisk/apps
make[1]: Entering directory `/usr/src/asterisk/apps'
Makefile:14: *** missing separator. Stop.
I looked at the makefile and i dont see anything glaring, but then
again it's been a long time since i wrote any code.
tried
i am very happy with my Zyxel P2000Wv2. the latest firmware solved all
the problems (there were some NAT issues.)
i'm running SER in front of asterisk. all good, except that it appends
the port to sip requests and i had to put config in SER to handle
that.
sometimes there's a huge echo, but i'm
hello all,
a slightly off topic question...i've successfully installed the AGI and admin and user interfaces of a2billing. everything seems to be working fine. thanks to areski for a very nice piece of software.
however, since i'm not familiar with the calling card industry, all the talk of
Hello all,
forgive me if this is a simple question, but does bridging a SIP channel and an IAX channel that use the same codec (say, alaw) involve transcoding? i'm trying to figure out what kind of hardware i'll need, and i'm going to be using SIP endpoints and IAX trunking to move the audio along
hello,
trace the SIP packets and see if they are actually addressed to 5062. if you post the ngrep or ethereal dump we'll see whats actually going on. I do this with SER on 5060 and asterisk on 5070 and there are no problems - my extensions point to 5060 and my DID's point to 5070 so asterisk
On 10/19/05, Yair Hakak [EMAIL PROTECTED] wrote:
i do it this way because i want all the dialplan logic and CDR having to do with PSTN in asterisk, not SER.
so, calls from the outside are adressed to [EMAIL PROTECTED]:5070 and hit asterisk. asterisk either sends them along to 5060, or handles
I dont know about others, but i find that SER as a SIP proxy in front of asterisk works much better for endpoints on the public internet than just asterisk as a sip proxy.
my 2 cents.
-yair
On 9/26/05, Federico Alves [EMAIL PROTECTED] wrote:
I want to share some facts with the Asterisk community.
go to pulver's blog, there's a free code.
-yair
On 8/23/05, Dean Collins [EMAIL PROTECTED] wrote:
Anyone able to get me a comp/highly discounted ticket to this?
$150 just to visit the exhibition halls sounds crazy?
Dean
-Original Message-
From: Jeff Pulver [mailto:[EMAIL
hello,
please 'scuse the slightly offtopic question, but i see a lot of
posts about the adit600, used as a channel bank, but from what i
understand it can be used as a PRI interface as well.
If anyone who is using the adit600 to interface to 4 T1/E1's has
feedback, i would appreciate it,
post your dialplan, it's pretty safe to say that's where the problem is.
without it, there's no way to help you.
-yair
On 8/16/05, Appan KH [EMAIL PROTECTED] wrote:
Hi,
I had configured Asterisk with the following
1). X100P - Card
2). Two -Greadstream100 SIP Phones.
I am able to make calls
exten = 198,1,Dial(SIP/198,20,tr)
exten = 198,2,Hangup
exten=_0.,1,Dial(Zap/1/SIP/197,20,tT)
exten=_0.,1,Dial(Zap/1/SIP/198,20,tT)
appan kh
- Original Message -
From: Yair Hakak [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
According to the CIA world factbook there are 800 million landlines in
use and about 6.4 billion people. This makes more sense than 800
billion. there are probably at least an equal number of cellular
telephones in use as well, but i have no idea how one would go about
getting those numbers
hi all,
is there a switch statement in the dialplan? or do i have to
daisy-chain GoToIf statements? i don't see a switch statement on the
wiki, but you never know...
thanks
yair
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
hello all,
i have a strange problemi am running SER in front of asterisk,
and am testing softphones.
x-lite works fine...i can dial, hang up, DTMF, all good.
Firefly looks really cool and i'm very impressed with the IM-like
interface and the skinning ability, but something strange is
Hello,
I have the following setup:
sip phones -SER - asterisk - voip provider1
- voip provider2
i got a toll-free DID from voipprovider1 to allow people from outside
to call into asterisk, get authenticated, and use voipprovider2 to
call out (kind
hello all,
I have a DID from nufone, transported via SIP to my * box, and even
though i'm using rfc2833 DTMF i'm still getting double digits and all
sorts of other stuff...
sip.conf is as follows:
[general]
port = 5070 ; Port to bind to
disallow=all;
hello all,
another desperate request for help debugging my dialplan...
from a certain extension i do the following:
DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM})
a NoOp to the console says
DBput: family=CFIM, key=2122022001, value=2122022001
and database show says
/CFIM/2122022001
the database with: /DB(CFIM/999) : 999 ?
What is the version of asterisk your machine runs?
Regards,
/* Ferdy */
http://asterisk.nsec.nl
info(AT)nsec(DOT)nl
Yair Hakak wrote:
hi ferdy,
i did check your first post to the list, and i really appreciate your help.
however
Hi list,
another question for you all, and i apologize in advance if it is
basic, the syntax is making me crazy and the documentation is no help:
when i do database show in the console, i get the following:
/DB(CFIM/999) : 999
and when i run the following
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's
phone). [Please reply through the mailing list]. Mike.
-Original Message-
From: Yair Hakak [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 5:05 PM
To: Mike Hillerbrand
Subject: Re: [Asterisk-Users] call forwarding, most basic case
hi,
thanks for your answer, but i'm
I believe this may solve your problem,
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone
works for me.
-yair
On 6/30/05, Betül Gözlükoğlu [EMAIL PROTECTED] wrote:
Hi;
Have a BUDGETONE-100 and using it with asterisk…Problem occurs when I dial
message
well, i can't say i'm surprised. any company whose approach to
customers is you are all scum trying to cheat us, don't ask
questions, and we'll help you when we feel like it isn't going to be
around for a long time.
On 6/26/05, Andres [EMAIL PROTECTED] wrote:
So it looks like Livevoip went
yes, there is.
run everything through asterisk, no matter how long the extensions
are. for example, 666 calls 999
goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER.
bounces back to ser. If everything is working well asterisk will set
up the call and get out of the way.
I
Hello Deepak,
1. don't post multiple times. it's annoying. enough said.
2. run asterisk in verbose mode (start it with asterisk -vgc),
place a call from a SIP endpoint behind SER to the asterisk server,
and see what happens in the asterisk CLI.
3. if you don't see anything there, get ngrep
Hello,
this is not automatic, you need to set up the proper dialing rules.
the fact that a DID dumps a call into the system and that there is an
extension with the same numbers do not mean they will be automatically
connected.
post your config files...
On 4/19/05, Nathaniel Angelo A. Torres
Hello Deepak,
yes, you can use mysql. the packages are in asterisk-addons.
there is a very good wiki page on the subject here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
hope this helps,
yair
On Apr 6, 2005 7:33 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
hi friends !
can
I suggest you strip naked, do a war dance, and sacrifice chickens to
the digium gods, and then i'm sure verything will work fine. If that
doesn't work, do the same thing while standing on your head.
Or, you could post some details of your installation so we have some
faint idea of what might
Jeez, people...learn to take a joke.
I offered to help the man. I didn't make cracks about googling or
anything like that. I said explicitly that if he posted details we'd
try to help. I didn't insult him, call him a worthness noob, or
otherwise offend him. IT WAS JUST A JOKE. Is this different
Hi,
a few pointers:
1. the wiki is your friend:
http://www.voip-info.org/wiki-Asterisk
lots of good stuff and good documents for getting started. If i
were you i might reinstall asterisk from CVS just to make sure you
have the latest version, and because this way you can learn about
Hello,
what is the benefit of your scenario #2? I'm not understanding what
it adds for you...
-yair
On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex [EMAIL PROTECTED] wrote:
Hi all
I have a couple of questions maybe you guys can help me with them
I have sip phones , SER server ,
Duh, i'm an idiot. I meant scenario #1.
-yair
On Wed, 23 Mar 2005 18:52:28 +0200, Yair Hakak [EMAIL PROTECTED] wrote:
Hello,
what is the benefit of your scenario #2? I'm not understanding what
it adds for you...
-yair
On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex [EMAIL PROTECTED
http://www.voip-info.org/wiki-Asterisk+billing
On Fri, 11 Mar 2005 19:58:37 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am looking at AMP and read All the graphic reports are based over
the CDR database.
How do I get the CDRs into a database?
bye
Ronald
the following is on voipjet's site:
Please note we are having a temporary glitch with our New York
location. Please send traffic to our West Coast Premium Server until
the problem is fixed sometime today. New SERVER IP: 69.25.60.30
although i guess an email to this effect would have been nice.
Hello,
i'm using ser+nathelper+rtpproxy in front of asterisk. It has been
terrific. The only problem i have is with some DSL modems that grab
port 5060 for themselves (why, i don't know, it's very annoying but
easily solvable). Other than that, no issues at all, in the NAT, in
the DMZ, between
See, here's the problem when you misrepresent yourself...the web is so
easy to search that any idiot like me can discover what you're doing.
http://lists.digium.com/pipermail/asterisk-users/2004-September/064464.html
'nuff said.
i'm sure their support is awesome. i'm sure it doesn't cost you a
and that was months
ago and it was a part of my freelance contribution.
Don't think others to have same kind fruadelent mentality that you
have.SO next time before proving yourself very smart from others take
the pain to ask what is the matter.
Ehsan
On Thu, 3 Mar 2005 14:09:06 +0200, Yair
hi,
you need to tell us how you're saving your cdr's - database, csv, whatever?-
if you're saving to a database a stored procedure is probably best,
unless you want to change the SQL statements in the proper module.
yair
On Thu, 3 Mar 2005 05:59:59 -0800 (PST), R A [EMAIL PROTECTED] wrote:
platform and you are jealous to
let know others about a good platform.
I think all the people here are matured enough to get their judgements
on the product rather than jsut ordering it because I said so.
Ehsanul Karim
On Thu, 3 Mar 2005 14:31:56 +0200, Yair Hakak [EMAIL PROTECTED] wrote
]
[mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak
Sent: Thursday, March 03, 2005 9:51 AM
To: M. Ehsanul Karim; asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re : Calling card platform
It is fine to tout your own products. we call that marketing.
However, anyone who claims
.
there are something else that you need to now??
wert
Yair Hakak [EMAIL PROTECTED] wrote:
hi,
you need to tell us how you're saving your cdr's - database, csv, whatever?-
if you're saving to a database a stored procedure is probably best,
unless you want to change the SQL statements
:[EMAIL PROTECTED] On Behalf Of Charles Wang
Sent: Saturday, February 26, 2005 11:50 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk + SER
Yes, I use this method too.
On Sat, 26 Feb 2005 18:18:15 +0200, Yair Hakak [EMAIL
you do not need radius for ser and asterisk to speak to each other. if
anything, i would suggest using SER for the endpoint and asterisk for
the billing and accounting.
-yair
On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote:
I just installed SER last night but if you want
ok, not that i'm such an expert myself, but
1. there's a big difference between newbies asking specific question
and the i want asterisk to run my life, make me coffee, and solve my
problems, does asterisk do that? questions that are appearing lately.
I'm not a member of the list police and they
Hi asterisk list,
this is a bit off topic, but can anyone explain the point of the
commercial far-end solutions floating around (jasomi, for example)? or
are the far-end things just hyped up media proxies? They claim to be
b2bua devices but that's a very wide category and only implies that
the
hi all,
does anyone know what time variables are fed to to the calldate
field in cdr_mysql? I have my system time set to israel time zone,
have restarted mysql and a show variables shows timzone as IST which
means now() should return israel time, but the calldate field keeps
getting the system
what endpoints are you using? You probably have a DTMF type mismatch
between asterisk and your endpoint (IP phone or softphone)
-yair
On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote:
Hello,
So far everything that I'm trying with asterisk is working except for this
i've actually had reboot issues since moving to 1.0.5.16, the phones
seem to hang more often on soft reboot and require a hard reboot
(unplugging). This is just a feeling and i can't quantify this but i
don't remember having to physically reboot the phones this often
before. I'm using one bt-101
hi list,
is anyone succesfully using asterisk with libretel port-of-call
(www.libretel.com)? If so, i would be grateful for configs..i set up
libretel to forward to [EMAIL PROTECTED]:5070 (asterisk is running
on 5070 and SER on 5060) and when i call the number i see SIP messages
with ngrep but
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the
] On Behalf Of Yair Hakak
Sent: Saturday, December 04, 2004 2:08 PM
To: Ian Chilton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail for Current Extension?
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have
Hello,
I know the d-link units (DVG-1120 ATA and their router as well) are
supposed to work well with asterisk...does anyone know if the units
that come with ATT callvantage are locked, or can they be used
w/asterisk or SER? And if they are locked, is it linksys no way out
locking or a simple
Hello,
try this document (from the wiki):
http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf
setting the auth param and the canreinvite and reinvite might help.
-yair
On Thu, 11 Nov 2004 10:38:55 -, Ashling O'Driscoll
[EMAIL PROTECTED] wrote:
Hi,
I havent received many replies
use the wiki, luke.
http://www.voip-info.org/wiki-Asterisk+Configuration+from+database
On Tue, 02 Nov 2004 07:16:09 -0600, Director General: NEFACOMP
[EMAIL PROTECTED] wrote:
Hi list. Does anyone know of any configuration that will make asterisk
read the extensions from a MySQL database
hi list,
anyone have any success getting asterisk to pass message waiting
indicator to a grandstream with SER in the middle as a SIP proxy? I
recently implemented SER between asterisk and my SIP clients and it's
significantly more stable (no more dropped clients) but i haven't been
able to figure
if by digital phone you mean IP phone like a grandstream or a snom,
then yes, you don't need any additional hardware to connect to *
(except an rj45 cable, of course.)
-yair
On Fri, 22 Oct 2004 10:19:18 -0400, David Ishmael
[EMAIL PROTECTED] wrote:
I'm sure this has been asked more times
Hello list,
please take a look at these units:
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596
are they locked? does anyone have one working with asterisk or SER?
Are these rebadged units from a different manufacturer?
anyone have any experience good
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure because anyone can bypass the SER
and register themselves as a peer with the asterisk. assuming
i've been playing with cdr_mysql and the Master.csv file, and since i use a
macro to define extensions the csv and the db both save the destination of
the call as s, instead of the destination.
macro is as follows:
[macro-extensip]
exten =s,1,Dial(SIP/${ARG1},10)
exten =s,2,Voicemail(u${ARG1})
: [Asterisk-Users] help a poor newbie out with SIP choppy
one-way problem
Date: Thu, 19 Feb 2004 11:53:17 +0100
From: yair hakak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Thu, 19 Feb 2004 07:54:00 +
Subject: [Asterisk-Users] help a poor newbie out with SIP choppy one-way
problem
Reply
Hello all,
i have a one-way choppy sound problem that i can't fix...
here are the relevant points
1. i am running 18.2.04 CVS on rhl 9.0 on a hosted server with a wide pipe
up/down with no hardware, just SIP connections and voicepulse for outgoing
IAX calls.
2. conecting to * with SJPhone (SIP)
Does anyone have any ideas on how to stop these messages from the SJPhone?
everything i've seen says they're harmless, but they're filling my console
and if anyone has any ides on how to make them go away i would be
appreciative.
thanks,
yair
is anyone using h.323 to send outgoing traffic to a voiIP termination
provider? if so, could you send me a sample h323.conf file and the relevant
line from extensions.conf
thanks-
yair
_
The new MSN 8: smart spam protection and 2
; Port to bind to
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
[yairphone]
type=friend
insecure=no
username=yairphone
secret=yairphone
host=dynamic
dtmfmode=inband
callerID = Yair Hakak
nat=true
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