On Thu, Aug 23, 2007 at 03:13:40PM -0700, bilal ghayyad wrote:
Dear Mojo;
Thanks for your help.
Why you said export ASTERISK_PROMPT=new prompt ?
To make that a new value for the environment variable.
An alternative method is:
ASTERISK_PROMPT=new prompt asterisk -r
There are actually
On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote:
Steve Totaro wrote:
David Gomillion wrote:
On 8/23/07, *Ed Pastore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me
I'm trying to build a test suite so that I can run calls through and
verify the call results.
I've made a cross over cable and linked my 2 ISDN30 ports together. So
now I can dial out on span 1 , and to receive the call on span 2.
in the context for span 2, I have the following:
snip
; #1
Hi
I Have been asked to setup a virtual call center. the server will be
hosted at the ISP, with the incoming lines for the local telco .. the
calls from the incomming lines will then be forwarded to individual
users directly or to other call centers ..
Any suggestions on web mangment interface
Oh, man, why is it that when you spend hours working on something, as
soon as you send a message for help, the solution presents itself ?
To answer my own question, and for prosperity, see the comments inline:
Sorry for the waste of bandwidth :(
Julian Lyndon-Smith wrote:
I'm trying to build
Dear all
I have now install TE210P 2 port E1 card on asterisk 1.4.10 on
centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back
and second E1 card on Direct Telcom for outgoing for outside now i got this
error when i call on avaya PRI
asterisk think
Hi.
Please help. When trying to compile Zaptel 1.4.5.1 I get the following:
/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c
base.c:48:29: linux/workqueue.h: No such file or directory
base.c:292: warning: `vpm150m_firmware' defined but not used
make[2]: ***
Dear Gordon;
Thanks a lot, it is working and it was from the
firewall.
But what is the command that I can know all the
rigestered endusers (iax2 or sip or h323)?
I tried iax2 show registry but it did not give any
thing? Can u help?
Regards
Bilal
--- Gordon Henderson [EMAIL PROTECTED]
wrote:
On 8/23/07, [EMAIL PROTECTED]
Hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
Hi,
you have to correct your etc/zaptel.conf as follows
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,0,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
then try
Regards,
Vidura
==
I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5
but thing
Dear All,
Am happy to say that I've successfully been able to register a SIP user from
a soft phone terminal via LDAP. The biggest hurdle that I had to overcome
was the LDAP-Asterisk schema. The schema example given in the astirectory
installation document is incomplete.
Here's are a few
Greetings everyone
I've set up a call recording system on debian 4.0 with asterisk and mysql db
for handling user information (accessible over the net for users). My
asterisk is running on one machine and the mysql on another. The connection
is over lan. Now I have a problem and a question.
My
You need to look at pri_cpe vs pri_net.
PaulH
On Fri, 2007-08-24 at 01:27 -0700, satish patel wrote:
Dear all
I have now install TE210P 2 port E1 card on asterisk
1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with
avaya E1 back 2 back and second E1 card
Hello,I am new to asterisk but i have vbeen scriptinh PHP SQL and webLanguages
for a long time.I can Give you a solution but using php
AGI:extensions.con- AGI(connect.agi);/var/lib/asterisk/agi-bin/connect.agi
:#!/usr/bin/php
Please see the official tracker in the Digium buglist:
http://bugs.digium.com/view.php?id=5768
Here are the schemas we did for OpenLDAP:
http://bugs.digium.com/file_download.php?file_id=14842type=bug
http://bugs.digium.com/file_download.php?file_id=14841type=bug
Also, for Novell eDirectory,
On Fri, Aug 24, 2007 at 08:29:55AM +, Jan du Toit wrote:
Hi.
Please help. When trying to compile Zaptel 1.4.5.1 I get the following:
/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c
base.c
base.c:48:29: linux/workqueue.h: No such file or directory
base.c:292:
Stephen Bosch wrote:
Ryan M. Colbert wrote:
I’ve had requests to processes incoming voicemails with voice
recognition routine and add the output text to the body of the email
message from * with the attached .wav file. Has anyone implemented this
type of feature and willing to share some
Hello ppl,
Sorry to re-post it, but kinda these issues are getting on my nerves.
I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on
1.4.4.
The problem :
1. I receive call from caller 'AAA' on my number, 'BBB' which is on my
Asterisk box.
2. I have to redirect the call to
here is the sip debug for the channel. Before reading the sip debug i want
ot tell you that user is using Telco Systems AC-211 v4.50.27 adapter. sip
sdebug shows that asterisk is trying to send the initial invite but there is
no response from the user (after registration, user dies, no single
Tzafrir Cohen wrote:
On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote:
Steve Totaro wrote:
David Gomillion wrote:
On 8/23/07, *Ed Pastore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while
Gregory Machin wrote:
Hi
I Have been asked to setup a virtual call center. the server will be
hosted at the ISP, with the incoming lines for the local telco .. the
calls from the incomming lines will then be forwarded to individual
users directly or to other call centers ..
Any suggestions
On Fri, 24 Aug 2007, bilal ghayyad wrote:
Dear Gordon;
Thanks a lot, it is working and it was from the
firewall.
But what is the command that I can know all the
rigestered endusers (iax2 or sip or h323)?
I tried iax2 show registry but it did not give any
thing? Can u help?
sip show
Thnk for reply
can you tell me one thing what is the meaning of this line
span=2,0,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
why did u use 2,0,0 ?
one more thing can you send me your config file i want to see more option
send me your /etc/zaptel.cong and /etc/asterisk/zapata.conf file
Dear all
I got one more error my asterisk E1 card connected with avaya
E1 card
[avaya]---E1-[asterisk]
i got this 2 error what is start asteris on consol mode
asterisk -c
[Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're
the CPE,
Search the wiki, Google it, you will feel better than being spoon fed.
It is all about timing on the E1. Who is providing the timing and who
is taking it. There is plenty of info on www.voip-info.org that will
explain this much more than a few responses on the user's list.
Thanks,
Steve
Set it to pri_net instead of pri_cpe. IF you start getting error
messages that We think we are NET and they think they are NET, then your
carrier or the Avaya has the line in Loopback mode.
satish patel wrote:
Dear all
I got one more error my asterisk E1 card connected
Andrew Kohlsmith wrote:
On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote:
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr)
Will this work even if the Local is pointing to a Zap channel?
As far as I know, this only works with SIP or IAX outgoing.
I'm not sure where you are getting
Stephen Bosch wrote:
That's not what was in your example. Your example is a mix of Zap and
SIP. Zap channels answer immediately, so if you do Dial() to multiple
technologies, the Zap() channel will always answer first.
This is not quite accurate; Zap channels that are analog FXO ports
answer
Just for starter, look at CallWave, and Jott.
-E
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I see that the term now is chan_mobile to use a bluetooth to cellphone
trunk. (what is in a name? :) )
What I want to know is:
Is there any restriction on the bluetooth chipset for the server?
Can I use the dongle for PAN and chan_mobile at the same time?
Can I use the dongle for headset (a
On Thu, 2007-08-23 at 18:42 -0300, [EMAIL PROTECTED] wrote:
1- I've tried running fxotune
2- I've tried turning off all un-necessary hardware in the BIOS
3- I've tried on a different PCI slot.
4- I've tried these suggestions:
On 8/24/07, EdPimentl [EMAIL PROTECTED] wrote:
Just for starter, look at CallWave, and Jott.
-E
They seem to be commercial :(
A quick search in google revealed a page with some compilation of
opensource STT engines.
http://www.faqs.org/docs/Linux-HOWTO/Speech-Recognition-HOWTO.html
Making
Hello Jared,
On 8/24/07, Jared Smith [EMAIL PROTECTED] wrote:
Digium offers installation support on their hardware cards, so if you
continue to experience problems, Digium support should be able to help
you track down the cause of the problem.
I also have the same issue on my TDM400P card
Stephen Bosch wrote:
Anthony Francis wrote:
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr)
Will this work even if the Local is pointing to a Zap channel?
As far as I know, this only works with SIP or IAX outgoing.
-Stephen-
I use local because It dials the call from default
snip
Until 1.4 improves, I'm staying with 1.2
/snip
Ditto
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Hello,
Every queue has some status counters (completed, abandoned, hold
time...) that are very useful for statistics. The problem is that those
counters are reset every time Asterisk restarts.
Is there a way to keep those counters, maybe in astdb? Also, is there a
way to reset the counters
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
Tzafrir Cohen wrote:
On Thu, Aug 23, 2007 at 07:16:57PM -0400, Lee Jenkins wrote:
Steve Totaro wrote:
I stay with 1.2.12 or somewhere around there. End Of Life but seems
to have a better ticker than 1.4.
Thanks,
On Fri, Aug 24, 2007 at 10:09:19PM +0800, GNUbie wrote:
Hello Jared,
On 8/24/07, Jared Smith [EMAIL PROTECTED] wrote:
Digium offers installation support on their hardware cards, so if you
continue to experience problems, Digium support should be able to help
you track down the
Hello,
I have two user machines, each with a jain-sip-applet-phone installed on it. I
use the following process to try to make a call:
1. Register each phone with the Asterisk server (working).
2. Add a contact in each phone which is the other user. (Get a 489 Bad Event
SIP error shown
Hello,Maybe I don't understand what DTMF in ASCII means but I can't make my
record stop using this syntax in a PHP agi script :fwrite(STDOUT, RECORD FILE
/var/lib/asterisk/ENR/jeanpaul wav '#' 15000 BEEP s=3000\n);The php syntax
isn't a problem because I really start recording, I have a beep,
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote:
Hello,
Every queue has some status counters (completed, abandoned, hold
time...) that are very useful for statistics. The problem is that those
counters are reset every time Asterisk restarts.
Is there a way to keep those counters, maybe in
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote:
Every queue has some status counters (completed, abandoned, hold
time...) that are very useful for statistics. The problem is that those
counters are reset every time Asterisk restarts.
Is there a way to keep those counters, maybe in astdb?
On Fri, 2007-08-24 at 03:44 -0500,
[EMAIL PROTECTED] wrote:
Message: 20
Date: Thu, 23 Aug 2007 23:13:55 -0500
From: Jay Milk [EMAIL PROTECTED]
Subject: Re: [asterisk-users] xPL and Asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
My decision based on what I've been reading in the bug tracker and
people commenting on how they've had
Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was
installed from ubuntu-server and asterisk loaded from source)?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is
This is the full log that I get after my trial run:
Aug 24 14:15:51 VERBOSE[3710] logger.c: -- Registered SIP '202' at
192.168.1.250 port 9810 expires 120
Aug 24 14:15:52 VERBOSE[3710] logger.c: -- Registered SIP '201' at
192.168.1.251 port 8529 expires 120
Aug 24 14:15:55 NOTICE[3710]
You can do it from svn server , I think there is a page in the wiki
Best,
yann
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jeremy Mann
Envoyé : vendredi 24 août 2007 17:30
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
finally found a solution for stuck channels. Im sharing this for anyone who
may face this problem in future. It seems a bug as i can repeat this
behaviour both ways as many times as i want. i will start an issue for this
on digium bug site but first i have to test it on the latest version of
While it is not exactly running a huge system, I have had one 1.4
system running in a small office of 10 phones since June with no
problems and another small system for about a month with no problems.
I have also had a larger system (80+ phones, DUNDi and IAX trunking to
11 sites) running 1.4 for
Doug Lytle wrote:
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
My decision based on what I've been reading in the bug tracker and
people
(sent to the list because it's pertinent)
Bilal,
anything can be a unix variable, it doesn't matter what it's called.
#!/bin/bash
export THREE=3
export FIVE=5
((EIGHT=$THREE+$FIVE))
echo $EIGHT
THREE, FIVE, and EIGHT are all just
I'd be interesting in pooling resources for this. We've seen the success of
Vonage's Visual Voicemail and would like to emulate a similar solution.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
What is a good softswitch that is also open source rather than asterisk?
On 8/24/07, James Jones [EMAIL PROTECTED] wrote:
Yes you could, but asterisk was designed to be a PBX. I would not use it
as
soft switch due its limitations. It really depends on how much traffic you
are going to be
On Friday 24 August 2007 12:37, Ryan M. Colbert wrote:
I'd be interesting in pooling resources for this. We've seen the success of
Vonage's Visual Voicemail and would like to emulate a similar solution.
Please define success,
I have a vonage account, and the transcription is very poor at
On 8/24/07, Joshua Colp [EMAIL PROTECTED] wrote:
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have in their specific scenario?
I do, but many of the problems I have
Joshua Colp wrote:
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have in their specific scenario?
I do, but with limited resources for testing (2 polycoms, no
James FitzGibbon wrote:
On 8/24/07, *Joshua Colp* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have
James FitzGibbon wrote:
Let me ask a question myself: what kind of regression test does * undergo
before release, and what level of traffic gets put through stuff like
app_queue? I assume it's not real-world scale, else these hard to pin down
concurrency issues we're seeing would have been
On Fri, 2007-08-24 at 22:09 +0800, GNUbie wrote:
I also have the same issue on my TDM400P card but I am not in the US
so I don't know how I can call to your number, 877-546-8963 for free.
You can call Digium at +1-256-428-6000 (which obviously wouldn't be a
free call, but at least you can get
svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
thegui; cd thegui; sh configure; make sudo make install ; clear ;
echo 'completed'
-bk
Yann JOUANIN wrote:
You can do it from svn server , I think there is a page in the wiki
Best,
yann
I understand this question is over-broad, but hopefully you can have
patience with a newbie and toss me a bone...
I am in the testing stage of deploying Asterisk. I have successfully
configured it to work behind the NAT of my ZyXEL ZyWALL 35 firewall.
However, I think there is a lot of
On 8/24/07, Joshua Colp [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
I'm going to end this email with
Joshua Colp wrote:
Doug Lytle wrote:
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
My decision based on what I've been reading in
I stay with 1.2.12 or somewhere around there. End Of Life but seems
to have a better ticker than 1.4.
Thanks,
Steve
1.2.12/14/17 all have seemed very stable to me so far.
Both of which are anecdotial evidences.
Now suppose I had a major stability
Still true on CentOS 5. You can only RAID partitions unless you do the LVM
thing. What are the disadvantages compared to being able to RAID the whole
disk? Maybe for monitoring it's just more to deal with but does it make a
RAID 1 any less reliable?
-Original Message-
From: Zane C.B.
It compiles fine for me but I can't change the soft EC. It always compiles
with MG1 no matter what I select in zconfig.h. Downgraded back to 1.4.4 and
it works fine again.
-Original Message-
From: Jan du Toit [mailto:[EMAIL PROTECTED]
Sent: Friday, August 24, 2007 1:30 AM
To:
We have been running 1.4 since July 06 (it was trunk then), and have
upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571).
We have 100+ extensions (SIP) and 30 ISDN channels. We often have 50+
agents available for outbound calls and queues (20+ queues). We are
making /
I'm pretty sure that a command to reset the counters was added soon
after this patch.
Julian.
James FitzGibbon wrote:
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote:
Every queue has some status counters (completed, abandoned, hold
time...) that are very useful for statistics. The problem
Russell Bryant wrote:
James FitzGibbon wrote:
Let me ask a question myself: what kind of regression test does * undergo
before release, and what level of traffic gets put through stuff like
app_queue? I assume it's not real-world scale, else these hard to pin down
concurrency issues
Doug Lytle wrote:
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
My decision based on what I've been reading in the bug tracker and
people
Steve Totaro wrote:
1.0 is over-ripe or rotten/forgotten and thrown away. Besides, we all
know that 1.0 was just a marketing ploy to legitimize Asterisk. What
serious company is going to install 1.BETA2 or .90? Maybe a
nonessential piece of software but not something as mission critical
If you still have this problem, contact me via MSN at the same address
I write from. Im sure that with 5 minutes in your box we can fix it.
Regards
On 8/24/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On 8/23/07, [EMAIL PROTECTED]
Hi,
I am using debian 4.0 with version
In article [EMAIL PROTECTED],
Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
We have been running 1.4 since July 06 (it was trunk then), and have
upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571).
We have 100+ extensions (SIP) and 30 ISDN channels. We often have 50+
Dear all
I am going to install asterisk on production now i m
confused about version i dont know which version is good and best for my setup
1.2.x or 1.4.x
can anyone tell me in detail which version whoud be best for my setup 1.4.x
or 1.2.x
if 1.4.x is good
Dear
which Linux version would be fine for asterisk CentOS 5.0 or
Debian 4.0 or RHEL 4.0
Regards
Satish patel
-
Choose the right car based on your needs. Check out Yahoo! Autos new Car
Finder
Thnk
now it is working fine according to your reply
Regards
satish patel
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Set it to pri_net instead of pri_cpe. IF you start getting error
messages that We think we are NET and they think they are NET, then your
carrier or
Asterisk Project Security Advisory - AST-2007-021
++
| Product | Asterisk |
|+---|
Hi
On Fri, Aug 24, 2007 at 03:08:56PM -0700, satish patel wrote:
Dear all
I am going to install asterisk on production now i'm
confused about version i dont know which version is good and best for
my setup 1.2.x or 1.4.x
Installing 1.2 on a new system now (that
Hello,
On 8/24/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Not sure about restarts, but trunk keeps them through reloads. How
often are you restarting?
My Asterisk has been segfaulting a few times during the day. I couldn't
figure out why that's happening. safe_asterisk restarts Asterisk
If I issue a restart gracefully command, the system will wait until all
channels are idle before restarting.
During the time the system is waiting for idle activity, is there a command
that can let me know it is in graceful restart wait mode ?
Thanks,
Ron
Hi List,
I have a 2Mbps SDSL link which gets saturated during peak time because
about I have about 3 E1 worth of g729 traffic going thru. So I'm planning
to use IAX2 trunking to reduce bandwith requirement and squeeze out each
and every bit of this (expensive) bandwith.
I've set up two
www.freeswitch.org http://www.freeswitch.org/
(still in early beta)
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Quitoriano
Sent: Friday, August 24, 2007 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Fri, Aug 24, 2007 at 12:27:14PM -0500, Russell Bryant wrote:
James FitzGibbon wrote:
Let me ask a question myself: what kind of regression test does * undergo
before release, and what level of traffic gets put through stuff like
app_queue? I assume it's not real-world scale, else these
On 8/24/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Hi List,
I have a 2Mbps SDSL link which gets saturated during peak time because
about I have about 3 E1 worth of g729 traffic going thru. So I'm planning
to use IAX2 trunking to reduce bandwith requirement and squeeze out each
and every
On Fri, Aug 24, 2007 at 04:00:23PM -0400, Matt Florell wrote:
With all of that said, I do have a testing setup that allows me to run
tests at high loads on Asterisk, but not all scenarios can be checked
in a testing setup. I ran a mid-volume test on 1.4.10 and it worked
without crashing. I
CentOS and RHEL are the same thing. One uses the RedHat trademark, the
other doesnt. One is expensive, the other isn't. I don't like to
recommend either because I just don't like RedHat's business
practices.
Personally I recommend SuSE Linux. OpenSuSE without the GUI installed
will do just fine.
Launched the OS X version of Gizmo after about a year of inactivity,
downloaded the update and discovered the new improved Giszmo features
Asterisk interoperability by allowing a secondary SIP account to be
registered simultaneously.
It also allows you to make the routing choice for
I have used CentOS and it works fine and it is easy to install. I know
that Debian is a little more complicated to install Asterisk and some
teatures on Debian.
I'd choice CentOS 4.2 or 4.4, as my personal preference.
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