I have seen two references today (don't recall whether here or on one of
the other VoIP lists I read) to people having the .78 version of the
firmware installed on their phones.
I'm keen on getting hold of it, but their support page still offers only
the older version.
Is it out there
On Wed, 20 Aug 2003, Scott Lambert wrote:
On Wed, Aug 20, 2003 at 12:13:07PM -0500, Dave Weis wrote:
On Wed, 20 Aug 2003, Mike Ciholas wrote:
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to
There was some talk on the IRC channel about getting a
convention/conference together for Asterisk users. While I do not
have the authority (or time) to make such a proposition, I'd like to
see if I can gather some support for a BOF (Birds Of a Feather)
meeting on a somewhat smaller scale and
OK, this thread is getting really out of hand, so I'll condense my
answers into one big stupid message:
1) 911 service. Yes, that is one of three reasons to keep your PSTN
line. The other two reasons are: Inbound calls from local callers
still should work on a POTS line, for now. You
Any chance of handouts, transcripts, or video being posted to your website
soon?
On Wed, 20 Aug 2003, John Todd wrote:
For those of you that are in the Portland, Oregon area:
I am giving a talk today on Asterisk at the PLUG Advanced Topics
Meeting. Details below.
JT
[snip]
Yes, I
OK, that VOIP dialtone? thread was getting really out of hand, so
I'll condense my answers into one big ugly message:
1) 911 service. Yes, that is one of three reasons to keep your PSTN
line. The other two reasons are: Inbound calls from local callers
still should work on a POTS line, for
Cheers mate!
After getting the latest CVS, I manage to get it work in my AGI script.
Excellent patch, thanks a lot.
Foong
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Actually it seems to depend on the brand of RJ-45 jacks.
I used to work in an office that routinely plugged RJ-11s plugged into RJ-45 jacks.
They had tested
various brand RJ-45 jacks and used those that were tolerent of having RJ-11's plugged
in.
However, I agree that its in general a bad idea.
One thing that PBX systems do well that VOIP systems mostly don't do at all well is
support
handsfree intercom/paging functions. Some phone systems will even let you page to
an in-use
extentsion. So if you depend on these kinds of functions, it may be hard to replace
them with
current VOIP
Hi all,
I have been asked to look into using asterisk as part of our setup.
The eventual goal is to replace as many parts of the existing setup as
possible, but in the interim, I just have to make it bolt on and work
with all existing parts.
My current setup is as follows:
Cisco 7940
(ext
Hello all,
A newer version asterisk-oh323 (v0.5.5) is available.
This version contains updates for compilation with latest
Asterisk sources, some additions (atexit cleanup)
and some other fixes.
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
Tebaldi Marco wrote:
Hi all,
i' using asterisk with chan_oh323 to test the SIP/H.323 interoperability. All working well...
Now i would like to know if is it possible that * doesn't control the RTP traffic.
I would tht the end point control the RTP flow...
Any idea, is it possible???
No,
Michael Bielicki wrote:
Did you get ilbc into it ? openh323 supports it as a pluggeable codec module
..
I have done some preliminary coding but it's far from stable,
so it's not in yet.
Michael.
On Thursday 21 August 2003 12:58, Michael Manousos wrote:
Hello all,
A newer version asterisk-oh323
Or you can just tftp01.sipphone.com and get it! :P
bkw
On Thu, 21 Aug 2003, Steve Meyers wrote:
On Thu, 2003-08-21 at 00:32, Brian Capouch wrote:
I have seen two references today (don't recall whether here or on one of
the other VoIP lists I read) to people having the .78 version of the
Hi All,
I've been developing all sorts of applications for use on our 79xx handsets but am
having great difficulty with formatting, I just can't seem to be able to produce a
line feed between lines on the stuff actually displayed on the phone. Has anyone else
has experience or success with
Your post implies that the non-voip implementations of asterisk support
intercom/paging. If this is so how do you get it to work?
Gene
-Original Message-
From: James H. Thompson [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 21, 2003 5:37 AM
To: [EMAIL PROTECTED]
Subject: Re:
Hello,
I am looking for a pbx solution which is not too expensive
but flexible :-) (a customer is in need of a call center
and crm solution).
The customer favors linux on the call center server and
the crm clients.
On a search for solutions i found Asterisk but it looks as
supports analog phones
Peter,
Did you read the website? Not only does it support h323.
Inter-Asterisk Exchange (IAX)
H.323
Session Initiation Protocol (SIP)
Media Gateway Control Protocol (MGCP)
http://www.asteriskpbx.com/index.php?menu=features
bkw
On Thu, 21 Aug 2003, Peter Eckhardt wrote:
Hello,
I am
On Thu, 21 Aug 2003, John Todd wrote:
1) 911 service. Yes, that is one of three reasons to keep your PSTN
line. The other two reasons are: Inbound calls from local callers
still should work on a POTS line, for now. You can't find VOIP
providers in most area codes
Packet8 looks like
Hello,
Did somebody test the Dialogic Hardware with
Asterisk:
- D/120JCT-LS
- D/41JCT-LS
If you can somebody tell me how it works with
Asterisk.
Thanks,
Bart
On Thu, 2003-08-21 at 02:31, Dan wrote:
Hi,
Which is the usualk way to do auto forward in Asterisk?
I need to be able to entert a number(code) from my phone indicating the new
phone number when I will be available.
Then when someone calls my old number, just the new one to ring.
This
Hi list,
I would like to know of a possible way todial
a pstn number with an extension .
Let the number is 56626965-234 so now i
wanna dial 56636965 then wait for some time and dial the extension 234 to reach
a particular person.I am afraid that i could not figure it out.
I am trying in
Thanks!!
Just added to the bug list as a request
Even with wide pipes , a few g711 calls , running without VAD , can really
fill a pipe.
Lee Goodman
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 4:41 PM
Subject: Re:
Mike Ciholas [mailto:[EMAIL PROTECTED] writes:
Oh well. I'm would expect no one would have presence here.
...
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D [EMAIL PROTECTED]
Hi all,
This is a NEWBIE question, so all you experienced types that are
tired of stupid questions can move on...
I've pretty much given up trying to do my entire phone system
over IP (including local service), so I have to select and
provision my local CO lines. I need about 10-12 lines
Hello,
I have download Xphone Lite and I cannot hear what
caller sys to me.
It is somethign with codec ?
Bart
Hello,
Thanks to members of the list things changed on my installation
(asterisk + digium E100P with of course an E1 line).
Here is my zaptel.conf :
-
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=fr
defaultzone=fr
-
Output of zttool:
-
Alarms Span
YEL/RED Digium
On Thu, 21 Aug 2003, Low, Adam wrote:
I've been developing all sorts of applications for use on our 79xx handsets but am
having great difficulty with formatting, I just can't seem to be able to produce a
line feed between lines on the stuff actually displayed on the phone. Has anyone
else
*CLI show application AbsoluteTimeout
-= Info about application 'AbsoluteTimeout' =-
[Synopsis]:
Set absolute maximum time of call
[Description]:
AbsoluteTimeout(seconds): Set the absolute maximum amount of time
permitted
for a call. A setting of 0 disables the timeout. Always returns 0.
On Thu, 21 Aug 2003, Mike Ciholas wrote:
However, while everyone can sell me POTS lines, when I ask about
getting these in some sort of digital muxed interface, I seem to
confuse the providers. In one case, I was able to get something
called channelized T1 which cost a lot and did not
Oh, and let's not forget that the traditional carriers are
not ignorant
of what is happening with VoIP or customer interest. There
is no doubt
that they are aware that if they don't find a way to deliver
this service,
someone else will.
No, if they don't find a way to deliver
On Thu, 2003-08-21 at 10:20, Mike Ciholas wrote:
Hi all,
This is a NEWBIE question, so all you experienced types that are
tired of stupid questions can move on...
Ah, but you followed all the right things to do, no HTML email, started
a new thread, and you didn't demand someone solve your
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Ciholas
Sent: Thursday, August 21, 2003 10:21 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Provisioning CO lines
Hi all,
This is a NEWBIE question, so all you experienced types that are
On Thursday, August 21, 2003 10:21 AM, Mike Ciholas
[SMTP:[EMAIL PROTECTED] wrote:
Hi all,
This is a NEWBIE question, so all you experienced types that are
tired of stupid questions can move on...
I've pretty much given up trying to do my entire phone system
over IP (including local
Nufone won't answer their phones. I am very interested in finding out
pricing from them as Jeremy stated they are very good with their rates.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Roach
Sent: Thursday, August 21, 2003 10:23 AM
To:
On Thu, 2003-08-21 at 10:58, Steve Meyers wrote:
On Thu, 2003-08-21 at 08:33, Steven Critchfield wrote:
BTW, what size is size=3D2 ? It seems to be in all HTML email from
Microsoft products.
http://www.ietf.org/rfc/rfc2047.txt
It's part of the MIME spec for encoding non-ascii text. =3d
WipeOut . wrote:
I have seen two references today (don't recall whether here or on one of
the other VoIP lists I read) to people having the .78 version of the
firmware installed on their phones.
I'm keen on getting hold of it, but their support page still offers only
the older version.
Is it
I'm brand new to asterisk but not to T1s so here's my bit to contribute.
Each local telco {be they ILEC or CLEC} is different depending on their
CO switch and the software options they've purchased for it.
In Alaska, the break-even for switching from POTS to T1 is about 13
trunks.
Your telco
Peter Eckhardt [EMAIL PROTECTED] writes:
I just found the draft of the handbook. The software is
amazing
Does anyone use Asterisk in Germany on a BRI (S2M) interface ?
http://www.junghanns.net/asterisk/page1.html
Driver to use * with a capi compliant isdn card. I currently use AVM
Mike,
I opted for an integrated T-1 for 1 customer who needed about 12 lines.
I configured it with 12 lines voices and 768k data. Chances are you need
this kind of bandwidth if you need 12 phone lines. Combining it on 1 T-1
can make it a little more cost effective and of course one of
On Thu, 2003-08-21 at 17:10, Peter Eckhardt wrote:
I just found the draft of the handbook. The software is
amazing
Does anyone use Asterisk in Germany on a BRI (S2M) interface ?
I'm in the Netherlands, but I use Asterisk on a BRI using a Fritz!Card
ISDN adapter and the chan_capi
Gabe Bourque wrote:
Hello Anton Tinchev,
I'm writing to you in hopes you can answer a few questions regarding
Asterisk/Digium and it's reliability. I saw your posting in the
Asterisk mailing list (Re: [Asterisk-Users] Is Asterisk ready for real
use?) and decided to write directly to you.
Hi all,
i would like to know if it is possible to bridging the rtp traffic over Asterisk...
I would like that the RTP flow is not controlled by * but by the endpoint.
Is it possible??? Any suggestion to do this???
Thanks
Marco
On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote:
Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards
are cheaper and has more features. Any ISDN card that is supperted by isdn4linux
must work, but I recommend you Sedlbauer chipset based.
Digium FXS cards are great.
On 20 Aug 2003 15:50:15 +0300, Johanna Kangas wrote
I understand the problem is in hisax. I am a woman, not a
stupid :)
So anyone who have had same kind of problem WITH HISAX ?
-Johanna
I, for one, haven't. However, I recommend moving away the whole
offending /lib/modules/2.4.x/
Eric Wieling wrote:
On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote:
Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards
are cheaper and has more features. Any ISDN card that is supperted by isdn4linux
must work, but I recommend you Sedlbauer chipset based.
Digium FXS
On Thu, 2003-08-21 at 20:08, Grzegorz Nosek wrote:
After
modprobe capi
modprobe fcpci
/proc/capi seems ok (shows one card with fcpci driver - sorry I don't
post some real output but I had to revert to i4l to make it work as
soon as possible)
So far, no error messages of any kind, but
Interesting development in Minnesota PUC regulating VoIP:
http://news.com.com/2100-1037_3-5066652.html?tag=fd_top
- Justin
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BRI (more correctly called ISDN BRI) is a digital service.
On Thu, 2003-08-21 at 14:12, Anton Tinchev wrote:
Eric Wieling wrote:
On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote:
Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards
are cheaper and has more features.
On Thu, 2003-08-21 at 20:38, Eric Wieling wrote:
BRI (more correctly called ISDN BRI) is a digital service.
That may be a technical answer.
On Thu, 2003-08-21 at 14:12, Anton Tinchev wrote:
Here Analog = BRI
That could be a financial answer?
--
Dave Cotton [EMAIL PROTECTED]
Check your codec.. I had that problem when trying iLBC.. G.711 and GSM work fine..
Hello,
I have download Xphone Lite and I cannot hear what caller sys to me.
It is somethign with codec ?
Bart
--
__
http://www.linuxmail.org/
Now with e-mail
I must put working 4 sales agents.
They will have PCs on the workplaces, so I thing that some Linux software phone with
headset is better solution
___
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[EMAIL PROTECTED]
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Yes, I'm familiar with the E911 platforms and their requirements to
some degree. The trick is that the people running Asterisk PBX
systems have no visibility into SS7, and that is an unreasonable
expectation, so some other out-of-band method for moving caller
location to the PSAP is required.
There are TDM interfaces higher end PBX's use to interconnect to the PS/ALI.
I beleive it's a CAMA trunk that signals using MF.
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 21, 2003 3:00 PM
Subject: [Asterisk-Users] RE:911, networks
Dave Cotton wrote:
On Thu, 2003-08-21 at 20:38, Eric Wieling wrote:
BRI (more correctly called ISDN BRI) is a digital service.
That may be a technical answer.
On Thu, 2003-08-21 at 14:12, Anton Tinchev wrote:
Here Analog = BRI
I mean the Price of course.
That could be a
In this message I try to summarize what I have learned in these last two weeks. My primary sources of informations were the * list archives and linux ISDN docs. I ain't no * master, so don't trust too hard.
Relevant messages from the * list for the current discussion are: 009177.html 009268.html
I discovered and deployed a solution some would consider counter-intuitive.
For whatever reason, I can get a dedicated long-distance T1 for about $400
MRC ($16 per line) while a local T1 costs over $1,200 MRC ($50 per line).
My telco automatically assumed I would want/need the local T1 for my
At 00:35 21-8-2003 -0500, you wrote:
On Thu, 2003-08-21 at 00:26, Chee Foong wrote:
Yes, I see that in the source code. But how do I know how many users are in
the conference room in real time.
I mean how can I retrieve the number of user from meetme? Do I need to edit
the source?
As of this
On Thu, 2003-08-21 at 14:38, David Carr wrote:
I discovered and deployed a solution some would consider counter-intuitive.
For whatever reason, I can get a dedicated long-distance T1 for about $400
MRC ($16 per line) while a local T1 costs over $1,200 MRC ($50 per line).
My telco automatically
On Thu, 21 Aug 2003, David Carr wrote:
I discovered and deployed a solution some would consider
counter-intuitive.
I love out of the box thinking. What kind of business is it?
For whatever reason, I can get a dedicated long-distance T1 for
about $400 MRC ($16 per line)
From who?
For
And also I cannot transfer a call from X-Lite phone but i can transwer from
ATA
Strange...
Bartek
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 21, 2003 4:13 PM
Subject: Re: [Asterisk-Users] Xphone Lite Cannot make work on Asterisk
Hi Pedro,
On Thu, 2003-08-21 at 21:34, pedro bulach gapski wrote:
My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2) running on
standard debian woody.
I'm not sure, but isn't there a linux capi driver available for that
card? If you, I suggest you try to use the capi
Has anyone had any issues with background noise while using a TDM400 card?
If so, what things did you tweak to resolve the issue? My * server has a
single TDM400 card (2 ports enabled) with two X100P cards.
Any feedback would be greatly appreciated.
bottom response = on
On Thu, 2003-08-21 at 15:17, [EMAIL PROTECTED] wrote:
Has anyone had any issues with background noise while using a TDM400 card?
If so, what things did you tweak to resolve the issue? My * server has a
single TDM400 card (2 ports enabled) with two X100P cards.
Any
Does anyone have a working example of how to use the switch directive
to peer two Asterisk PBXes?
--
- Ian C. Blenke [EMAIL PROTECTED]
(This message bound by the following:
http://www.nks.net/email_disclaimer.html)
___
Asterisk-Users mailing list
How are people handling call transfer with SNOM phones? We are okay with
the # transfer workaround, but I worry about how that will work with
other systems that expect me to be able to press # to return to the
previous menu or similar.
Thanks,
--Ernest
On Thu, 2003-08-21 at 15:26, Ernest W. Lessenger wrote:
How are people handling call transfer with SNOM phones? We are okay with
the # transfer workaround, but I worry about how that will work with
other systems that expect me to be able to press # to return to the
previous menu or similar.
Hi,
In testing the Budgetone we have noticed something strange with DTMF and
Voicemail. When we set the Budgetone for RFC2833, and connect to voicemail,
the detected DTMF digits do not correspond with what we pressed on the phone.
For example user=1001, password=1001 is detected as:
Incorrect
- Original Message -
From: Michiel Betel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 19, 2003 11:53 AM
Subject: RE: [Asterisk-Users] CDR-Event on AstManager
The manager inteface currently sends the following events with the
associated parameters:
Event: Newexten Channel
I have not had any problem at all with the 10 I have.
They sound good and work well. The only problem I
ever had was a problem with remote ntp servers.
Andres wrote:
Hi,
I would like to know if others have experienced a high percentage of Budgetone
defective units. We purchased 4 to test
Steve,
I pay 2.9 cents a min inbound 800 and outbound. Email
[EMAIL PROTECTED] I think he is being overloaded with requests for
information. It takes him all over 30 seconds to set someone up.
bkw
On Thu, 21 Aug 2003, Steve Lane wrote:
Nufone won't answer their phones. I am very
marrandy wrote:
Mine (2-weeks old) came with 1.0.3.77.
According to the web site http://www.grandstream.com/y-service.htm
It's still 1.0.3.77 and trying the tftp at 4.3.153.56 doesn't yield any
improvement. (I havn't sniffed the network yet to see if it's tftp is
working yet though).
gee, this sounds worse than any current virus or worm.
has this vunerability been reported to the appropriate NON cisco people
so a worldwide alert can be issued ?? :-)
On Wed, 20 Aug 2003 11:27:39 -0700, John Todd wrote:
For those of you wanting to salvage your Cisco ATA-186 after
WipeOut,
Thanks a bunch. I just saw this after I wrote my other e-mail thanking everyone for
their help.
I think I am going to DL RH 9 and just re-vamp my box with this install model. I had
planned to
dedicate my * box anyway eventually. Once again, thank y'all a heap. I'm from SE
OK, in
I love out of the box thinking. What kind of business is it?
Market research call centers - but no telemarketing :)
From who?
Originally, with a regional (western states) carrier called TelAmerica.
Later, I made myself the customer of record for a DS1 local loop to a major
central office
Which brings me to an application at hand:
We currently have an * box connected to a SIP Media Gateway which is
connected to the PSTN via SS7.
We have MF FG-C 911 trunks connected to a DMS and can bring an MF T-1 into
the * box if we buy a T400P card.
The question is how do you support
I am having problems using the manager even though
I am following the instructions from the Manager.rtf doc.
In manager.conf I have the
following
[general]
enabled=yes
port=5038
[fred]
username=fred
secret=fred
read=system,call,log,verbose,command,agent
This is excellent data; thank you. I'll review before harrassing the
E911 folks at VON.
However, this only seems to solve problems for local PSAP
connectivity to your * server. Or am I mistaken?
I think a larger and simpler (read: less hardware and no custom
circuits) system needs to be
At 04:37 PM 8/21/2003 -0600, you wrote:
I neglected to mention that we also maintain a POTS line in each office for
outbound 911 calls and to route one local call at a time to save on LD
charges where we can (because I'm such a tightwad). Asterisk makes it easy
to route 7 digit dials
Hi Howard,
My * server has a brand new 350 watt power supply. After your message, I
did confirm with Digium that the next batch of TDM400 cards is supposed to
resolve the issue.
Thank you for all your assistace.
Jeff Gunther
[EMAIL PROTECTED] wrote on 08/21/2003 04:20:16 PM:
bottom
Our phones have been working perfectly fine all day. I've personally
supported quite a few new users over the phone today and even set a
couple up.
Jeremy McNamara
Steve Lane wrote:
Nufone won't answer their phones. I am very interested in finding out
pricing from them as Jeremy stated
NUFONE R0X!
Took him 30 seconds or so to set me up when I got services with him! :P
bkw
On Thu, 21 Aug 2003, Jeremy McNamara wrote:
Our phones have been working perfectly fine all day. I've personally
supported quite a few new users over the phone today and even set a
couple up.
Jeremy
Hello All,
Sorry about the html, I always send mail using plain text, not sure why it
contains html.
Yes I should patch my outlook :).
My purpose is to limit the conference call for 1 hour. After that all
callers involve in the conference will be disconnected.
AbsoluteTimeOut hangup a
I'm not sure my configs would be of much use as we have 40 asterisk servers
with one unified extensions.conf so it is approaching 4000 lines, has 500
global config variables, and about 3 dozen macros/menus.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Hi all,
I like to have a dial in modem on my toll free number so that
when I or my employees travel, they can always get in for net
access to read email if no better method is available.
Right now, my Panasonic KX-TD1232D PBX receives the call on a
POTS line and routes it to an analog modem.
On Thu, 2003-08-21 at 21:24, Chee Foong wrote:
Hello All,
Sorry about the html, I always send mail using plain text, not sure why it
contains html.
Yes I should patch my outlook :).
My purpose is to limit the conference call for 1 hour. After that all
callers involve in the conference
On Thu, 2003-08-21 at 19:06, Ernest W. Lessenger wrote:
At 04:37 PM 8/21/2003 -0600, you wrote:
I neglected to mention that we also maintain a POTS line in each office for
outbound 911 calls and to route one local call at a time to save on LD
charges where we can (because I'm such a tightwad).
As far as people are thinking of sharing their telephony, we could let
people start exposing them through iaxtel. If anyone has areacodes
prefixes they want to make available, e-mail me and I can set you up on
iaxtel.
Mark
On Thu, 21 Aug 2003, Brian West wrote:
NUFONE R0X!
Took him 30
From the thread.
Subject: Re: [Asterisk-Users] IAX IAX trunking... DP cache?
Date: 20 Aug 2003 11:29:59 -0600
From: Steve Meyers [EMAIL PROTECTED]
Brian West wrote:
I would use the latest CVS for one. And try again.
Unfortunately, I've tried numerous times to get a current CVS trunk
Uh oh. I think I may be looking at the wrong tool.
My goal is to implement (in an open source software suite) an RTP/UDP/IP
header compression algorithm that would save bandwidth used by voice traffic
packets. So a 5ms G.711 packet that would otherwise be 98 bytes, could be
reduced to 62
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