Any ideas? Using background dosen;t work, because you hit # and it
hangs up.
I think you have to define a # extension in your macro, something like
exten = #,1,Playback(not-available)
exten = #,2,Goto(somewhere)
If I'm wrong, please someone correct me
hth
Can somebody tell me where to buy Sangoma ISDN cards in different
countries or is there a list of local resellers somewhere?
Why don't you ask them directly
http://www.sangoma.com/company/contact.htm
They should know where to direct you
___
Is asterisk 2.0 real? Running in c#? I see references to it but cannot
find it anywhere.
You must be referring to this post :
http://lists.digium.com/pipermail/asterisk-users/2005-April/098601.html
Please note the date of the posting, and this particular phrase in it
: Asterisk 2.0 was
I can modprobe TDM400P card without feeding power
supply , so what is the purpose of providing power
supply in that card.
Can any body tell me
It is needed if you have FXS ports on it, because the card will need
to provide ringing voltage to the phone.
hth
I use and recommend wavepad.
I using it also, it's a great free tool. You can find it here :
http://www.nch.com.au/wavepad/
hth
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To
Has anyone found a suitable but not exorbitant 4-6 port FXO = sip
gateway? I need something more compact than a channel bank and using
many Sipura 3000s is a bit cumbersome. Suggestions?
Mediatrix makes a 4 port FXO to SIP device :
Do you use this? Are you happy with it's performance? My experience a
year ago with other small FXOs was very dissappointing. I tried the
very earliest SPA-3000, TDM-400 and X101p. None were satisfactory so
resorted to call forwarding my two POTS lines to a lines from an ITSP.
Nope, I don't. I
As I recall, should channels start as channel=2 and not channel=2?
I have all mine config'ed channel = 2 and it works fine...
Greg
Yes, thanks for the correction.
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Ok, so the scope of the configuration is from channel= to channel=
statement with the configuration for the channel coming before the
channel statement.
As in...
these=are
configs=for
the=first
channel=1
these=are
configs=for
the=second
channel=2
In fact, all the settings that
If anyone is interested I'm (slowly) developing a GPL'd Java applet that
works as an IAX softphone.
I should have a test version out at the end of the week for a
limited number of testers.
I would love to be a beta tester on this
___
I have not been receiving mail from the list 29th July, what is the problem
with gmail and the list.
No problem here.
Check you Spam folder, and if you find email there from this list,
select them all and click Not spam
hth
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Asterisk-Users
I would like to setup an option in my digital receptionist that callers
can select to hear a read back of their Caller ID. It would be something
like, the number you are calling from is I think I can reuse the
festival script that is built in, but ideally this could be accomplished
Using recent Asterisk stable, we have a situation where I have an analog
door phone which, when the button is pushed, simply off hooks the line (no
auto-dial) and acts like you just connected tip and ring. In our previous
According to the specs of this device, you can program it to dial a
what is the best web based managment aplication for asterisk ???
This topic as been discussed many times. Search the archives :
http://www.google.ca/search?hl=enq=site%3Alists.digium.com+web+management
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Can anyone recommend a good soft phone that's easy to configure under
Asterisk and works well on a typical Windows XP system?
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
If you want a SIP phone, try X-Lite :
http://www.xten.com/index.php?menu=productssmenu=xlite
When calling from a DIAX to a IAXY, using ulaw, I can hear a thing AT
the DIAX, but the IAXY CAN hearme...
What the @@!#$Y@!#$!$%?
Any advices is welcome!
Make shure that your IAXY config (in iax.conf) only allow ulaw
like :
disallow=all
allow=ulaw
hth
I'm new to this matter and I spent three days in trying to connect one SIP
Softphone to an Asterisk Box. I always get error 401 or 403...
I don't understand very well settings in Softphone program: con anybody show
me how to set up a minimal running system with no public lines or external
I am facing some problem with DBdel it says Error deleting key from
database. Please let me know whats going wrong.
here is the syntax what iam using
exten = _*73.,1,DBdel(CF/${EXTEN:3})
Just a guess but, maybe the key does not exist ?
___
This is software. Use manageble software. If software means separate
setup on each desktop, then don't use it. If you spend that much time on
setting up phones, imagine how long it takes you to update other
software packages. This is, then, a symptom of a general problem.
I would like to
exten = s,9,System(curl http://127.0.0.1:13370/cgi-bin/sendsms?
username=namepassword=passto=12122122121from=12122121212text=Message
+text+here+${CALLERIDNUM})
Change it to this and it should work :
exten = s,9,System(curl http://127.0.0.1:13370/cgi-bin/sendsms?
OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can
receive and place calls with no issues, however, when I receive a call, the
CID
only shows Analog Line on the Grandstream 2000XP phone. Does anyone have
any
ideas even where to look to change this?? Is it a
Any idea if this applies to Mac OS X clients? We are a strictly Mac
company, and OS X's Unix core allows for preemptive multitasking. If
I am unhappy with the performance of the soft phones, I should be
able to tweak the priority of the phone so that it gets more compute
cycles.
I don't know
1.) User interface - The interface to the softphones is really less
than ideal. This includes the problem mentioned earlier about not
hearing ringing unless you have your headset on, dialing with the
mouse, not having telephone service if your PC isn't on, etc. The
traditional telephone
Can anybody recommend an Asterisk GUI to help a newbie confg ?
If you really need a GUI that makes it simple, try [EMAIL PROTECTED] :
comes with AMP.
See http://asteriskathome.sourceforge.net/
If you only want a web-base GUI to help you edit config files, follow
instructions on this page :
Hi again, folks. I've been getting feedback from this list and
elsewhere that softphones are generally not considered good enough
for hardcore business use. Can someone point me to where I can find
more detail on this debate?
Search the list. there is been a lot of talk on this subject.
Try
*CLI iax2 show peers
Name/UsernameHost Mask Port Status
z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored
z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored
From this, you can see that none of the IAX
What about define those phones on the SIP.conf and use sip, instead of IAX.
That protocol use be more used to communicate Asterisk servers more than
phones.
That's not totally true. An IAX softphone will work easily behind a
NAT/Firewall. The same can't be said for a SIP one. I've tested IAX
Hi all,
I need some help/guidance on writing the specs needed on a project
that will be scaling up to 10,000 users.
I will have some T1's to provide PSTN connectivity, and all the users
will be SIP and/or H323 phones. Services offered will include
conferences, voicemail (20 megs per users), etc
Jon, etc..., the issue here is her family all uses
special features of SBC voicemail. E.g. Her mom
leaves a VM for my wife at 10:30 PM after the baby is
asleep without fear of ringing the phones and possibly
waking up the baby. They use it like email. They
The only way I see is make an
The sound set I had was Canadian French. Appuyez sur la touche carré
type messages.
If you know of other French sound sets, please let me know!
I would be interested in the French-Canadian set, where can I get them ?
From a French-Canadian view, the french prompts from sineapps sounds VERY
Hello
Maybe a silly question, but after some searching couldn't find answer. Is
there a number I can dial to pickup and listen to my voicemail messages on
my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages
on Avaya system?
If you are using [EMAIL PROTECTED],
On 6/18/05, Balaji NJL [EMAIL PROTECTED] wrote:
Hi All,
I am a new bee to *. I just installed [EMAIL PROTECTED] on
FC3. I hv a FXO card. I hv configured two extensions
one x-lite and other iaxComm. I configured * using
AMP. The following setup works
- x-lite (x 200) to iaxComm (x 201)
-
On 6/20/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
snip
Is there anyway to shorten that or even give users the option to not play
that?
Voicemail as some options that can help you with that.
See http://www.voip-info.org/tiki-index.php?page=Asterisk+config+voicemail.conf
It can be
Take a look at the Asterisk Management Portal at
http://sourceforge.net/projects/amportal
It has a flash-based panel that will give you what you are looking for.
No need to install AMP to get this, just install FOP : http://www.asternic.org/
hth
On 6/14/05, Matt [EMAIL PROTECTED] wrote:
Hi,
What do I need to do to get asterisk to NOT pickup a Zap channel when
it rings? The channel in question is used for outbound calls only,
and all incoming calls are answered by an analog phone elsewhere in
the building that does not run through
On 6/14/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 14 June 2005 14:38, Time Bandit wrote:
[from-pstn-noanswer]
exten = s,1,Wait,3 ; Wait 3 seconds, to get callerid
exten = s,2,Hangup
No. You will get multiple entries then if they stay on the line and it rings
again
2. I have started to use realtime and have the hard disk with the mysql
data.
How can I use them now on the new machine.
In fact, I would like to use it on my database server directly, that
would help me to add two Asterisk boxes to one database.
You can just copy the MySQL files to the
Does anyone have any ideas on how to build an interactive IVR where
questions are asked by Asterisk (pre-recorded prompts), the caller answers
the questions, and the system records the answers and emails the whole
question-answer session as a .wav file? Similar to Comedian Mail except an
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Most people already have it : Quicktime
hth
___
Not entirely sure, but, I wonder what would happen if you define RING
in the [globals] section first, and the use SetVar or SetGlobalVar in
the other contexts to override its value.
Be aware that each call as his independant variable space. So if you
SetVar for a call, it doesn't affect a
You can run an X client on the station and display it in a remote X server.
Running X on an Asterisk box is, I think, a dumb idea. Numerous time
on this list it's been said to be a no-go
You can also edit files remotely via sftp (don't use ftp :-) ). You'll
find many familiar environments that
I'd like to known what I have to do to upgrade
the firmware into a IAXy device.
It does it automagically when it connect to Asterisk if a newer
version is available.
Look in /var/lib/asterisk/firmware/iax and you will see iaxy.bin.
hth
___
How can I manage my Asterisk using the web or somehow, since there are
too many configuration files and too many variables
You may take a look at [EMAIL PROTECTED] - it includes AMP (Asterisk
Management Portal) and other tools.
But it will be hard at the beginning, anyway :)
Or, if you
don't know what it (QoS= Quality of Service) is. I hope you may help me
giving Links to read and briefing me your ideas.
1 minute of google search I found this :
http://compnetworking.about.com/od/networkdesign/l/bldef_qos.htm
which looks like a pretty nice explanation
hth
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the
software and drivers, I am wondering if anybody knows where I could
downloaded from.
Have you tried http://www.mediatrix.com/ ?
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Can you point me to some basic doc on how to do that? I'm rather
familiar with linux (about ten years worth), but have never tried to
flop kernels like that.
First time I compiled my own kernel I followed the instructions on
this page : http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list
of callers to be blocked. When they call, they should hear busy and
then we hang up. We have about 100 DIDs routed to different contexts
and I wouldn't want to have to
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
If
I have 2 Gnet SIP phones connected on the same switch as the Asterisk
box. So far, our phones authenticate with *, because when I do sip show
users, I see our 2 phones there.
When you say that you see them, does it look something like this :
501/501172.16.1.201 D
Unfortunately the otherwise excellent Areski stat tool doesn't seem
to include the unique ID function and thus I can't pull a file back
directly from that tool
If you want Areski stat to store the unique ID, you have to modify the
makefile as explained in the installation document on the
How can I have asterisk ignore incoming rings so it doesn't answer a
specific line. I tried setting up an empty context section but that didn't
work.
What I did is this
[incoming-line-noanswer]
exten = s,1,Hangup
Works perfectly
Or, if you want your CDR to have the callerid, do it like
I have a local telco line hookup to my FXO port.. receive calls just
finebut when I try to dial out, it ring the phone connected to the FXS port
of the same card.not sure where to start looking to fix the problem, thanks
in advance for your time
You have other ports on this cards, at least
Can anyone recommend any free IP SoftPhones that are maybe open source?
Mine is not open source, but it's free for non-commercial use. Give it a try
http://www.marccharbonneau.com/asterisk/mediaxphone.php
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What is the best client's protocol for my softphones in Windows pcs? and
what is the best way for connecting clients, I meant should I use one
protocol for all the clients or some mix (SIP/IAX/oh323) of protocols?
what
is better?
I would say IAX. If you only use 1 protocol for
-- Executing Dial(SIP/3001-e13a, ZAP/1/65869804) in new stack
This is what's wrong I think. The line is missing the 'g' for the trunk
group. On all of my [EMAIL PROTECTED] boxes the cli shows
-- Executing Dial(SIP/227-a4dd, ZAP/g0/3428463) in new stack
It depends how you set it up
Is there any way to modify this text coming from comedian?
Yes, look in voicemail.conf
hth
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Hi List,
What is the good client softphone for windows that connects to my Asterisk
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
Give it a try
N.B.: I should release an updated version really soon now
___
Asterisk-Users
We are needing a IAX softphone with support for supervised transfer
(among others). We have tried the following ones without success,
either because it does not support supervised transfer, or because
some other codec issue that made it unusable:
iaxComm
Diax
IaxPhone
IAXTelefon
Do
I have.
I keep going to the Spam folder, select them and click Not spam
Maybe if we all do this, it will fix the problem
Hoping we can keep using gmail !
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More bug fixes. *69 works now. Cisco stuff works. Lots
of other fixes.
is phpconfig fixed ?
when editing a file, it doesn't show the list of sections, it only list Header
What needs to be modified :
In the function OC_readConfFile around line 131 change :
$this-_OC_the_file[] = fgetc($file);
Someone please tell me there is another way to provision an IAXy other than
this horrid method.
http://www.digium.com/downloads/Iaxy_Installation_Guide.pdf
http://dacosta.dynip.com/asterisk
hth
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I currently use another PBX system which takes care of VM. Is there a
way to prevent [EMAIL PROTECTED] v0.6 from picking up Incoming calls?
I'd still like to dial out from Asterisk (I have IAX trunking on). Is
there a way to do this? My knowledge of the Extensions.conf is
limited.
Go in
Is there a search engine for this mailing list ?
Yes, and it's the best search engine in the world.
Just go to google and type this :
site:lists.digium.com somesearchstring
hth
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I have a Linux Fedora 3 Asterisk only box (2 FXO 2 FXS ports) with no
GUI or WEB server running. I can get to it remotely using Putty but I
want to add the capability to at least do Dial Plan configuration via a
browser. Do any of the GUI based configurations support such a setup. I
really
On those incoming calls on my IAXy I hear the other party on my IAXy,
But this other party can't hear me (the audio that's beeing sent from
the IAXy to asterisk can't be heard)
Does anyone have any idea what I can do about this?
Check your account settings in iax.conf for you IAXy. This is
I'm new to the VOIP world and need some advice. I currently have a premium/
full functioned Panasonic PBX installed in my house/ small office... and
have some extra unused telco lines available on the PBX. I'd like to use
one of these extra lines for VOIP into the PBX/ phone arrangement.
I am curious just what the advantages of an email forum over an online
one.
Main advantage : I don't have to go to a website and check if there is
some new post. I receive them when there is one.
I can't imagine that all the people on this list would go there every
day to check it.
Some
I want to know how to do a consultative transfer on the second call
I.o.w if a call come in,A and another call come in B and B asks to be
transfered to exten 200,I want to speak to 200 1st and the transfer B to
200.
Easy. Park the call, call B and talk to him and tell him where the
call is
[cellphone]
exten = s,1,Flash
exten = s,2,Dial,Zap/2/9729796243
exten = s,4,Congestion
I never done this, but I believe you are missing a final part.
If you do the same thing on a regular phone, the scenario would be this :
1- you are connected with the remote person
2- you hit
Is there a way I can (for sip users) also include say my [dial-911]
[dial-local] and [dial-longdistance].. bearing in mind that I want to
have different sips allowed to do different things so I can't just do
includes for those in my from-sip-internal.
Just make different context for different
What I am looking for is a small switch with QoS that I
can stick in ahead of the dsl modem. Plug in one connection
from the voice lan and one from the data lan.
Linksys WRT54G supports QoS, 4 LAN ports and a WAN port that can do DSL/cable
hth
___
Sorry about the previous post. Is this still available? The main
thing is I need a management tool I can use in commercial sales.
Yes, it is. And in fact I've made a couple improvements :
- you can restart Asterisk : now, when convenient, gracefully
- you can reload : SIP, extension
Why not just return the Linksys to vonage, get your 40 bucks back, and
buy the sipura spa2000? They are the same thing, also sold under the
vegastream brand.
Or, why don't you return it to Vonage, get your $40 bucks, then buy a
P2P-NA from eezeephone.com at $59 + $10 shipping ?
For only $30
Why not just return the Linksys to vonage, get your 40 bucks back, and
buy the sipura spa2000? They are the same thing, also sold under the
vegastream brand.
Or, why don't you return it to Vonage, get your $40 bucks, then buy a
-NA from eezeephone.com ?
N.B.: the site is down now, the server
5) MWI, Call Waiting, 3-way calling missing
If I remember correctly (only used an IAXy a couple of times), it uses
shutter-tone to tell you when there's a message waiting
It definitely support Call Waiting : just use Flash as with normal
call waiting on the PSTN
Never tried 3-way calling, but I
There's another feature request. Let me dial ### or something to find
my IP...
That's not something to do with the IAXy, you can make an AGI script
that will tell you your IP. I had this script somewhere but I can't
find it at the moment. This would not only be valid for the IAXy, but
for any
So how am I going to provision the device in the first place, to be
able to dial this extension, if I don't even know the IP?
Oups, sorry, didn't think about this one.
Check winiaxyprov, the version 1.01 can scan your network to find
IAXy. Now the only thing we need is for Digium to write the
so seems like the verdict is go IAXy with a IAX only network ? Most of
the problems of the IAXy device seems like will be fixed with firmware
updates and wont require a hardware update..
The best part is how you update the firmware : each time an IAXy
connect to Asterisk, it check what firmware
Where do you put the module load and init commands on a RHEL 4 box and
where to put it on a RHEL 3 box?
If you are talking about loading the Zaptel modules, here's the easy way :
- Go in your Zaptel src directory (usually /usr/src/zaptel)
- # make config (this wil copy the init script)
N.B.: if
They probably don't want to deal with the snickering and laughter that
this code will ensue. Digium has a good rep. in the open source
community for Asterisk, they don't want to release this mockery on
anyone!
I truly hope your not right. While I know the code might be bad, that
What would be a good combination to use on dialup connections? I know
iax is better than SIP, but I dont' know much of anything about the
various codecs. Also, how well would an iax or sip solution work
compared to skype as far as voice quality?
I have a relative that is on dialup and
To have asterisk ignore incoming calls on an X100P fxo interface, do I
have to just not configure it in zapata.conf, or is there a way to
have the call ignored in a dialplan?
Just define a context like this :
[home-incoming]
exten = s,1,Wait,1 ; Wait 2 seconds, to get callerid
exten =
Callmanager does nothing than construct and tear down calls and the actual
RTP stream does not flow through the Callmanager but is direct from IP
device to IP device. How does this work with Asterisk? I read something
that lead me to believe that Asterisk has to process the entire call, is
[200]
type=friend
username=richard
change this to
username=200
And my X-lite Default SIP Proxy config is as follows:
Enabled: Yes
Display name: richard
Username: richard
Change this to
Username: 200
and this one
Authorisation User: richard
to
Authorisation User: 200
and it should
has anyone managed to get IAX client (firefly 3rd party version)
to work,
where the *Server is behind single NAT,
with port forwarding enabled on the NAT router, and
the client behind double NAT ?
clients behind single nat to * work fine.
Strange, I tested with iaxcomm and this was the
For SIP incoming/outgoing you normally need ports 5060 and the port
range 1-2 open.
At least it works in my setup.
Could anyone correct it if it's not exactly all the truth?
Just a little note that all these ports are for UDP, no TCP ports are used.
Would IAX phone work instead??
SIP should work, but it's not easy to configure.
And YES, IAX would definitely work behind NAT. That's the main point
of IAX superiority over SIP
hth
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receive calls from any of the 2 windows machines. The first windows phone
I start I can send/receve calls the second one I cannot. I. No matter
which one I start first only the first one works. The linux kphone can
Please take note that each phone need it's account. You can't have 2
phone
However now I cannot even browse a .conf file via phpconfig. When clicking
on the file I get the following error:
Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission
denied in /var/www/phpconfig/cls_phpconfig.php on line 127
I have gone over the wiki page, done chmod
We have also been looking at various GUI's for Asterisk... ([EMAIL PROTECTED]
being one)... can anyone recommend one that would be ideal for a business
user in a basic small / medium office environment?
Depends on what you mean by GUI
Simple GUI to edit/view Asterisk's config :
- phpconfig :
No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have
apache2-mod_php installed?
If you see the source of the script, you don't have php installed or
configured correctly.
go read this : http://www.php.net/manual/en/install.unix.php
hth
you and everyone else :-)
From: Daiku [mailto:[EMAIL PROTECTED]
But i AM looking for info on another IAX capable device - like the
IAXy, but more user
friendly, as it were...
Never bought from them, never played with the stuff, but check them
out anyway : http://www.iaxtalk.com/
hth
=== mod_php4-4.3.10_1,1 conflicts with installed package(s):
php4-4.3.10_1
They install files into the same place.
Please remove them first with pkg_delete(1).
Looks like you installed php as an AGI, not as a module of apache.
Remove it and install mod_php.
Fatal error:
I have a version of asterisk running on my server for more than 1 year. I
wanna update it to the latest version without over-writing any of the
config files.
How can I do this?
just don't type make samples when you rebuild it
___
Asterisk-Users
I was looking on the internet and couldn't find any link to install this
Mozilla extension.
http://www.sysnux.pf/cps/sections/telephonie/copy_of_mozphone/switchLanguage/en
Is it also possible to install it on Firefox?
Yes
Give it a try, it's a nice piece of software.
hth
even worse i cant login to my window manager being KDE as root :) it
doesnt permit root logins :) here we go.
Login as normal user, then open a terminal, type su, enter root's
password then type mozilla or firefox depending on what you have
installed. Then go install it
hth
[EMAIL PROTECTED] root]# mpg123
High Performance MPEG 1.0/2.0 Audio Player for Layer 1, 2 and 3.
Version 0.59g (97/04/23). Written and copyrights by Michael Hipp.
The recommended version is 0.59r
To download it, simply go in your asterisk src then type make mpg123.
Then, to install it go in
BTW: Whats actually that SendDTMF ? thing ?
http://www.voip-info.org/wiki-Asterisk+cmd+sendDTMF
DTMF definition : http://en.wikipedia.org/wiki/DTMF
N.B.: please try to trim your answers, the message is becoming pretty long
hth
___
Asterisk-Users
Questions:
1) Am i using an older version? If so, where can i get a newr version?
2) Am i missing some configuration, which one?
See this newly created document, it explains everything you need to
make it work.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig
It's been
Change [mateo01] to [1000] in your sip and you will be saying that ext.
1000 is registered with the specifics you are using.
Update the settings in your softphone to register the name and number as
1000
Then any attempt to dial 1000 should come to that phone.
Wiley
After doing thoses
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