use groups, check the commands/functions group and checkgroup.
On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
I suppose incominglinit=1 in the sip.conf of
I have installed oh323 channel driver (finaly! :)). I head some problem
starting * so I have put the smallest possible oh323.conf file to se what
happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts
but he also disables h323 channel because there are no available
Benoit Panizzon wrote:
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
I suppose
C F wrote:
use groups, check the commands/functions group and checkgroup.
I guess I can see how this would be useful, but is there no way to get
it to return BUSY in DIALSTATUS var?
On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote:
For multiline phones how do you set SIP channels to
Why not use the busy command, in combination with the groupcheck
commands - refer to http://www.voip-info.org/wiki/index.php?
page=Asterisk+cmd+Busy
On 09/04/2006, at 5:01 PM, Miles Scruggs wrote:
C F wrote:
use groups, check the commands/functions group and checkgroup.
I guess I can
On 04/08/06 11:26 [EMAIL PROTECTED] said the following:
I'm pleased to announce the release of Astmanproxy 1.20, the fast,
flexible proxy server for Asterisk's Manager Interface. Astmanproxy
we've just started using astmanproxy, and i'll soon be submitting a couple
of patches which
Dear Asterisk users,
I m
working on a final year research based project on Asterisk ... the work I would
like to take from Asterisk is to have voice conversation between two PCs
connected with eachother on a LAN with no Internet connection by using minimum
hardware ... plz if anyone can
Hello,
I posted a lot of mails may be asterisk is not able to
accept sip calls from internet !?
My english is not fluent i try my best !
My problem I use ser+asterisk.
For local calls there are no problem (PSTN or IP)
Now i wish to receive calls from other internet domain
but asterisk ask
On 04/08/06 11:26 [EMAIL PROTECTED] said the following:
I'm pleased to announce the release of Astmanproxy 1.20, the fast,
flexible proxy server for Asterisk's Manager Interface. Astmanproxy
we've just started using astmanproxy, and i'll soon be submitting a couple
of patches which
On 11:08, Sun 09 Apr 06, [EMAIL PROTECTED] wrote:
Hello,
I posted a lot of mails may be asterisk is not able to
accept sip calls from internet !?
My english is not fluent i try my best !
My problem I use ser+asterisk.
For local calls there are no problem (PSTN or IP)
Now i
On 9 Apr 2006, at 06:04, Bartosz Wegrzyn - asterisk wrote:
Anyone knows hot to fix that?
Thanks
I used to have my iaxy registered to my old version of asterisk.
I switched to 1.2 ver and now registration fails.
my config for iax.conf for that client looks like this:
[user]
username=user
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
${DIALSTATUS} BUSY comes from the phone.
On Sunday 09 April 2006 08:46, Benoit Panizzon wrote:
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up
Hi,
Sorry for my delay writting here. My SIP.conf is similar of yours, i
only don't use qualify=yes, is it compulsory? I have 100 users and if
i activate qualify it will increase the traffic in my network no?
Best regards,
Marco Mouta
On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
If
On Saturday 08 April 2006 20:18, Colin MacMillan wrote:
Hello,
6) From here I enter the qozap directory. cd qozap
7) now I get the following error -
linux:/usr/src/bristuff-0.2.0-RC8q/qozap # insmod qozap.ko
insmod: error inserting 'qozap.ko':
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
KhojaS wrote:
Dear Asterisk users,
I m working on a final year research based project on
Asterisk ... the work I would like to take from Asterisk is to have
voice conversation between two PCs connected with eachother on a LAN
with no Internet connection by using minimum hardware
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
Steve Totaro wrote:
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the
Grandstreams are totally useless, I had to switch all my phones to Linksys. Grandstream will not even support you and their router side do not work for the 486 or 496.
-- Original message -- From: "Andre Rodrigues (Cheyenne)" [EMAIL PROTECTED] I have more than 20 ATA
Miles Scruggs wrote:
Steve Totaro wrote:
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
Many multi-line phones allow you to use the same username/password for
all lines. Then the phone only actually registers once using that
username and password, not once for each line.
What we do with the Polycoms is configure each line to register as a
different username/password (we use the
Hi,
I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension for voicemail. Can anyone shed any light?
Thanks,
Waldo
Thanks for the help!
What I have gathered mentally so far is that asterisk can't do
exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts
from multiple peers ( 3 of them in sip.conf) from one single provider.
What happens is that it matches
Hi,
If you don't specify a host= statement in sip.conf and you have a
section that includes a username and secret plus type=peer, it will
match on username and secret. (That implies that if you have three
different numbers registered with your sip provider all under one
username, calls for all
Snip..
Thanks
Miles
If you type modprobe zaptel modprobe ztdummy at the Linux CLI,
what do you get?
Nothing, they were loaded before, and loaded just fine.
lsmod Module Size Used by
ztdummy 2608 -
rtc10620 -
I have installed oh323 channel driver (finaly! :)). I head some problem
starting * so I have put the smallest possible oh323.conf file to se what
happens. When I don't put available codec's in oh323.conf (*1) Asterisk
starts but he also disables h323 channel because there are no available
Not true. There are hundreds of thousands of
Grandstream adapters in use around the world. Grandstream support is not
perfect, but it is as good or better better than most vendors, including
Linksys/Sipura.The Grandstreams do currently have a bug with header
compression right now that
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
On Sun, 09 Apr 2006 09:12:42 -0700
Miles Scruggs [EMAIL PROTECTED] spake:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2)
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension for voicemail. Can
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could
Dear folks,
I got a problem sending faxes using spandsp. Primerily, when the tiff file
made using GIMP 2 with different compresions the fax app break downs whole
*. Moreover when i made a tiff file using Microsoft mdi, everything works
fine but on the other end of the call, the received fax
Snip..
Thanks
Miles
If you type modprobe zaptel modprobe ztdummy at the Linux CLI,
what do you get?
Nothing, they were loaded before, and loaded just fine.
lsmod Module Size Used by
ztdummy 2608 -
rtc10620
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
When you use groups you shouldn't even execute the dial command, but
instead use the busy command.
On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote:
C F wrote:
use groups, check the commands/functions group and checkgroup.
I guess I can see how this would be useful, but is there no way to
Hi,
I have download the uplink and test with skype 1.4 2.0. not lucky to me.
Only connect on first call then hang. I need to reboot my windows xp
everytime.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon
Sent: Wednesday,
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
Michelle,
you sent a single message containing suggestions to me on 11/02/2005.
Your claim to have contacted me many times is clearly false. Due to
demands outside the asterisk world, I have not been monitoring the list,
but I doubt that should have been necessary, considering that contact
Thanks Gavin.
On Uplink i have another kind of problem: the signalling is ok, but when i
try to answer my skype give me an error on audio part.
It could depend of nat or justabout port not open on my firewall?
Thanks
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL
kevin ling wrote:
Hi,
I have download the uplink and test with skype 1.4 2.0. not lucky to me.
Only connect on first call then hang. I need to reboot my windows xp
everytime.
Skype is evil. I would recommend you find a way to spend your time more
productively.
B.
Right, but it's asking for a user id not a number to dial. So, how
would I set it to dial extension ?
Thanks,
Waldo
On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one
Disallow=all
allow=ulaw
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
StrelnikovSent: Saturday, April 08, 2006 7:25 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Force
codec
Hi, Is it possible to force using codec depends
on
I just installed the script, it seems to hang while going out to the
web. Is there someway to have it run in the background while a
background() is playing or something like that?
Thanks
Miles
Jay Milk wrote:
Michelle,
1. Courtesy would suggest that you would have contacted the author of
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten = 100,4,Macro(hangupcall)
so the user doesn't need to put in a
Thanks
Waldo
On Apr 9, 2006, at 2:19 PM, Tim Litwiller wrote:
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten
Kerry Garrison wrote:
Disallow=all
allow=ulaw
N.B. the problem is depending on extension, not context or protocol. . .
B.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Michael
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could
Jay Milk wrote:
Michelle,
you sent a single message containing suggestions to me on 11/02/2005.
Your claim to have contacted me many times is clearly false. Due to
demands outside the asterisk world, I have not been monitoring the
list, but I doubt that should have been necessary,
in the sip.conf
insecure=very
canreinvite=yes
[]'s
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc: asterisk-dev@lists.digium.com
Sent: Saturday, April 08, 2006 11:41
Subject: [Asterisk-Users] 407 proxy authentication
Hello,
look at this I
Ok works like a charm now. So now when I dial ext , I get this:
Created MeetMe conference 1023 for conference '0'
So my question would be, how do I get other people to join this
conference? The voice prompts only tell me that You are entering
conference number X where X is 0,1,2
An Asterisk box at customer site shows these messages pretty regularly. This
causes one way voice, the called party cannot hear the calling party.
Apr 7 11:59:44 WARNING[18406] channel.c: Avoided initial deadlock for
'0x817b790', 10 retries!
Apr 7 14:47:46 WARNING[18406] channel.c: Avoided
Snip..
you could try dialing from another phone and to dial
either 1023
or 0, my guess is 1023 is what the other people will have to dial.
I would assume that it would work like that, but nope.
from a different phone just creates a new conf, and 1023 is
never announced it
Snip..
you could try dialing from another phone and to dial
either 1023
or 0, my guess is 1023 is what the other people will have to dial.
I would assume that it would work like that, but nope.
from a different phone just creates a new conf, and 1023 is
Hi All,
Not sure if this is a phone problem or an Asterisk problem.
Basically after a period of time (around 30 minutes but not too sure of the
time) the phone no longer delivers any sounds. What I mean by that is.
if I pick up the phone after a reset I get a dialtone. After around 30
Hi Everyone,
Things seem to work fine (except my phone audio issue in a previous mail)
I can leave a vmail message and it emails it out fine. However when I dial
the vmail server from any phone it usually resets the phone half way
through. There is no single point where it starts to do this,
Paul A Brown wrote:
Hi All,
Not sure if this is a phone problem or an Asterisk problem.
Basically after a period of time (around 30 minutes but not too sure
of the time) the phone no longer delivers any sounds. What I mean by
that is.
if I pick up the phone after a reset I get a
hi Ronald,
i would use a CallerIDNum authentication, based on the Asterisk Database to solve it.
then you do not need any verification.
you just build a list of approved numbers in the database and then have a context checking the whitelist.
if you need more help, let me know,
Mickey
On 4/8/06,
Miles,
I think this is a limitation of the AGI - I don't believe that asterisk can
fork a new process. If so, that would be interesting!
The script uses Wget - I believe we can set a timeout so that your system
doesn't hang waiting for the HTTP response. Let me know if that would solve
your
In both SCCP and SIP loads, dialtone comes from the phone itself, so if you're not getting that, it's probably a firmware problem. Does it affect all three sound systems (speaker, headset, handset) or just one of them?--JW- Original Message From: Paul A Brown [EMAIL PROTECTED]To: Asterisk
Hi
Its a SIP image, fairly old but seemed to of been ok in the past (I trashed
asterisk a while back and recently rebuilt it)
P003-07-3-00
I tried upgrading ot the latest but my dialplan.xml didn't work anymore
Thanks
Paul
- Original Message -
From: Jon Farmer [EMAIL PROTECTED]
Do you have a sccp config example I could look at
Thanks
- Original Message -
From: Jon Farmer [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 09, 2006 8:49 PM
Subject: Re: [Asterisk-Users] Cisco 7960
Paul A Brown wrote:
Do you have a sccp config example I could look at
http://www.voip-info.org/wiki/view/SCCP-HOWTO2
Regards
Jon
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I have had the exact same problem last week. I have not yet solved it.
So instead I am using ooh323, but would prefer to use oh323. Can anyone
help?
I'm glad that I'm not the only one :))
Hopefully we'll find solution to
What about different extensions using different connections?
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 10, 2006 4:26
Miles Scruggs wrote:
Ok works like a charm now. So now when I dial ext , I get this:
Created MeetMe conference 1023 for conference '0'
So my question would be, how do I get other people to join this
conference? The voice prompts only tell me that You are entering
conference number X
Hello Everyone.
I have a question, I have set up [EMAIL PROTECTED]
and I have purchased an ATA from TigerDirect. It is an MTA-102 from VOIP Solutions,
when I check the website that is in the documentation it is in Portugese from Brazil. I have done some investigation and it
seems that
Hate to reply to my own posting but I wonder if anyone know the answer?
Steve Totaro wrote:
Is there a setting somewhere in * to define whether I am receiving
callerID or true ANI? Global Crossing claims they are sending ANI but
I dont think so. My understanding of ANI is that it is always
When I register to a remote Asterisk system using IAX2, I can see it
notifying my Asterisk box that I have voicemail waiting. How can I get
Asterisk to use that information and send WMI to one or more of my SIP
phones?
Thanks,
Eric
___
--Bandwidth and
I want to make it global.On 4/10/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:What about different extensions using different connections?
Paul HalesTechnical ManagerAsteriskIT- Original Message -From: Brian Capouch [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial
I made an edit to the wiki:
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
While I need this solution, and I think that some other people can
benefit from various aspects of it, can anyone see if there is a more
elegant solution to achieve the same result? Please feel free to edit
I meant dialled extension, not originating
extension.
like :
exten =
_37X,1,Dial(IAX2/FAX/${EXTEN})
exten =
_38X,1,Dial(IAX2/NOTFAX/${EXTEN})
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
Michael
Strelnikov
To: Asterisk Users Mailing List -
There is two entry on the username which is the number assigned to the
port and the username apper on the sip entry u should put the same on to
it.. The number assinged is the authuser
Jerry Geis wrote:
I have a new wellgate 3802 unit. I have not gotten it to
register with asterisk 1.2.6.
Hi all,
My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any relateddocuments or weblinks?-- Thanks Best Regards!
Steven Li
___
--Bandwidth and Colocation provided
In my remember. The uplink install a virtual sound card. So uplink can auto
answer the call from skype or sip side and redirect to another side. No
matter what kind of onboard audio card do you have.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Hi All,
Could someone send me a code frag on how to get a record from the
asterisk database into a PHP variable via the Manager API?
I can issue calls, etc. from Manager. But I'm not comprehending how to
manipulate database variables.
Thanks much.
Darren Ellis
Zhiqiang Li wrote:
Hi all,
My client softphone supports IM feature. Does any warmheated expert
know if Asterisk can support IM also at server side? If so, is there
any related documents or weblinks?
--
Thanks Best Regards!
Steven Li
I am not sure exactly what you are trying to do
I tried the latest version of Jive over the weekend and I have to say it is
a giant pile of crap. I did this on multiple machines on both Linux and
Windows, and after setting everything up, the moment you add the asterisk
module, all authentication and user setup is lost and there is no way to log
Hi
Sorry for my chinese engish first.
The VIP-320 seems like a SIP ATA+DECT Phone product. Please check you have
registered to the asterisk server first. Because the VIP-320 built-in
H.323/SIP dual mode. The web config tool not so clear. Reference the
Page.31. Input the SIP:asterisk_ip_address
But in this case you have to define two users on both sides. It is not most likely.On 4/10/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I meant dialled extension, not originating
extension.
like :
exten =
_37X,1,Dial(IAX2/FAX/${EXTEN})
exten =
_38X,1,Dial(IAX2/NOTFAX/${EXTEN})
Hi,I have a single PRI span setup at present and need to dial a prefix number in order to suppress outgoing caller ID.I am about to have a second PRI Span set up on the same server, but I want to bring both spans into the one group. The second span will be from a different telco.
I want to
I have Jive (wildfire) 2.4.4 running on a win2k3 server box with the
asterisk plugin. Installed without significant problems, has been up
and running for about 6 wks now. Conference rooms especially are
convenient.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Mark Edwards wrote:
I have a single PRI span setup at present and need to dial a prefix number
in order to suppress outgoing caller ID.
Really? Normally you would set the calling presentation to 'restricted'
on a PRI, no prefix would be needed.
___
I have an issue with trying to ensure that when dialling an extension
that it continues to ring up to the timeout value. But what I am
finding is that the timeout is all over the place. Sometimes half the
timeout value and other times within a few seconds of the timeout value.
I am running
10 apr 2006 kl. 04.02 skrev Jolly M. Recto:
There is two entry on the username which is the number assigned to
the port and the username apper on the sip entry u should put the
same on to it.. The number assinged is the authuser
The username setting is not needed in this configuration at
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