Displaying 20 results from an estimated 39 matches for "x201".
Did you mean:
201
2004 Dec 18
5
Q about IAX (and IAXy)
...r just use the "received from" IP
address and port to respond?
Finally, would an IAXy work seamlessly in a configuration where it is
plugged into a NAT router which is plugged into another NAT router -
double NATted? The * server is on a public IP.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net
2004 Jun 16
6
Invalid Extensions -- More like traditional PBX systems?
...tones. I would
rather like this better than just hearing a busy signal on my phones.. I
DID search around on the wiki and using google and could not find anything.
Thanks.
--
Stephen Rosebush,
srosebush@desynched.org
http://www.desynched.org/
// Hardline // IP Phone
USA: 1-248-724-4452 x201 FWD: 63420 x201
Netherlands: +31-(0)20-6598858 x63420 x201 IAXTEL: 1-700-356-6191 x201
United Kingom: +44-(0)870-3403054 x201 SIP: sip:srosebush@desynched.org
2004 Dec 22
1
Asterisk billing solution
...ing that can handle monthly fees and per call charges
(depending on destination, obviously), and should provide a web
interface for customers and administrators.
Something that can tie in to one of the existing management GUIs would
be a big plus.
Any ideas?
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net
2004 Jun 22
1
Asterisk -- PBX Do Not Disturb
That could explain why it wouldn't work on any of my sip extensions I
tried it on this morning when I first read about it and thought cool the
things you learn.
Is there anyway to make it work on Sip extensions?
Cheers,
Dean
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J.
Angel
Sent: Wednesday, 23
2011 May 13
6
Powerful PC to run R
Dear all,
I'm currently running R on my laptop -- a Lenovo Thinkpad X201 (Intel Core
i7 CPU, M620, 2.67 Ghz, 8 GB RAM). The problem is that some of my
calculations run for several days sometimes even weeks (mainly simulations
over a large parameter space). Depending on the external conditions, my
laptop sometimes shuts down due to overheating.
I'm now thinking abou...
2019 May 15
1
domain still running although snapshot-file is deleted !?!
...254,14 335609856 183599 /mnt/snap/sim.sn (deleted)
CPU\x200/ 27007 27308 qemu 15ur REG 254,14 335609856 183599 /mnt/snap/sim.sn (deleted)
CPU\x200/ 27007 27308 qemu 16ur REG 254,14 335609856 183599 /mnt/snap/sim.sn (deleted)
CPU\x201/ 27007 27309 qemu 15ur REG 254,14 335609856 183599 /mnt/snap/sim.sn (deleted)
CPU\x201/ 27007 27309 qemu 16ur REG 254,14 335609856 183599 /mnt/snap/sim.sn (deleted)
vnc_worke 27007 27321 qemu 15ur REG 254,14 33...
2015 Jul 01
4
Dovecot auth username mapping
...I was hoping that was a possible replacement for this, but my goodness it was so incredibly slow! This would definitely be an option though, as it does serve the purpose. I just can?t figure out how to fix the performance issue. Any thoughts to this?
~ Laz Peterson
Paravis, LLC
Ph: 951.319.3240 x201
> On Jul 1, 2015, at 3:24 PM, Axel Luttgens <axel.luttgens at skynet.be> wrote:
>
>
>> Le 1 juil. 2015 ? 04:38, Laz C. Peterson
>
>> a ?crit :
>>
>> I have an interesting case here ?
>>
>> Virtual mailboxes, domain/username/aliases stored...
2015 Jul 02
1
Dovecot auth username mapping
Peter,
Yes that is a possibility. I will try disabling PAM (or switching the auth order) and see if that makes a difference. Thanks for the suggestion!
~ Laz Peterson
Paravis, LLC
Ph: 951.319.3240 x201
> On Jul 1, 2015, at 11:34 PM, Peter Chiochetti <pch at myzel.net> wrote:
>
> Am 2015-07-02 um 01:41 schrieb Laz C. Peterson:
>>
>> I did attempt to switch the PAM/Kerberos authentication to Dovecot
>> LDAP authentication, but now performance is unbelievably slow...
2004 Dec 17
0
Total newbie here looking to do a VoIPconference call?
...reports.com VoIP forum - there are several postings
(including a few by me) with instructions on how to downgrade the
DTA-310 to v1111, put in the SIP settings and upgrade to (not higher
than) v1234. I haven't tried it with *, but I assume it should work.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists...
2004 Dec 18
2
External Address Books
I'm not sure if this is possible, but I was hoping to find an address book
that runs on Windows XP that will allow me to select a phone number and send
that to my Asterisk. The Asterisk system would make the call and connect
the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out
there that can do that?
Thanks,
Dave
-------------- next part --------------
An HTML
2004 Dec 20
1
Problem using SPA-2000 behind NAT
Hello all,
I have a new Sipura SPA-2000 that I am trying to configure beind a
NAT. The SPA is able to register to the asterisk server without a
problem and the SPA can make calls to other extension that are not
behind a NAT. However, when I try to call the SPA from another
extension, the extension connected to the SPA rings, the user at the
SPA answers, and there is no audio in either
2004 Dec 23
3
error starting asterisk
Just upgraded to the current stable ver. when I start asterisk with
-vvvvvcg I get the following error
[pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined
symbol: pbx_substitute_variables_varshead
Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module
pbx_loopback.so failed!
Asterisk
2004 Dec 16
4
Polycom SIP Phones
Could someone please direct me (via personal email) to a provider with
good prices on Polycom Soundpoint IP 500's with POE cables? I need 14
of them.
Thanks,
Adam
________________________________
Adam S. Robins
Executive Vice President & CIO
PHARMACENTRA, LLP
5901B Peachtree Dunwoody Road, Suite 380
Atlanta, GA 30328
Office: 770-395-0088 x34
Fax: 770-395-0989
Mobile:
2004 Dec 18
1
One-way audio with SIP client only on certain calls
...real" IP, not a private IP. So, for some reason, STUN
(or something else) is seeing their IP as their ISP's router. It almost
seems like their connection is double-NATted?
I am unsure what steps to take next, so any help would be greatly
appreciated.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net
2004 Dec 17
2
Total newbie here looking to do a VoIPconfer ence call?
...dreports.com VoIP forum - there are several postings
(including a few by me) with instructions on how to downgrade the DTA-310 to
v1111, put in the SIP settings and upgrade to (not higher
than) v1234. I haven't tried it with *, but I assume it should work.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net
2015 Jul 02
0
Dovecot auth username mapping
It?s actually unbelievable how much slower LDAP auth is than PAM. Does anyone have any suggestions how I can improve Dovecot LDAP auth? I have tried caching authentications and that doesn?t help either.
~ Laz Peterson
Paravis, LLC
Ph: 951.319.3240 x201
> On Jul 1, 2015, at 4:41 PM, Laz C. Peterson <laz at paravis.net> wrote:
>
> Thank you for the response Axel. I will look into that.
>
> I did attempt to switch the PAM/Kerberos authentication to Dovecot LDAP authentication, but now performance is unbelievably slow. For e...
2012 Aug 06
2
redirect actions exceeds policy limit
...fo_log_path =
lda_mailbox_autocreate = no
lda_mailbox_autosubscribe = yes
log_path =
mail_plugins = quota sieve
postmaster_address = postmaster at rockisland.com
sendmail_path = /usr/sbin/sendmail
}
Mike Greene
Rock Island Technology Solutions, Inc.
San Juan Islands, WA. 360-378-5884 x201
2004 Dec 17
5
Total newbie here looking to do a VoIP conference call?
I am looking to help out my company find a more budget conscious but
reliable way to hold conference calls between 5+ people. 4x a month we hold
several hour long conference calls during non-business hours. All of the
employees have high speed internet. Currently we dial up an AT&T conf using
regular analog phones.
I don't have a great grasp as to what Asterick is capable of, but my
2004 Dec 17
14
Call on hold disconnects...
G'Day All,
How do I fix this:
I receive a call at the extension. Press the hold button. Music on hold
starts. When I place the handset back on the cradle, the call gets hung
up/disconnected. The Phone is A GrandStream Budge Tone 100.
Thanks
2015 May 12
2
Why is the diag function so slow (for extraction)?
>>>>> Steve Bronder <sbronder at stevebronder.com>
>>>>> on Thu, 7 May 2015 11:49:49 -0400 writes:
> Is it possible to replace c() with .subset()?
It would be possible, but I think "entirely" wrong.
.subset() is documented to be an internal function not to be
used "lightly" and more to the point it is documented to *NOT*