On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass a...@p2ee.org wrote:
we upgraded to 1.8.13.1 and we have much the same problem although after
the upgrade I don't seem to find any cases where the qualify value is
OK (xx ms
This has come up before on the list and archives but I don't seem to
find a solution for this. On just a few nodes we have this situation
where we see the IP disappear from the CLI iax2 show peers list but
the status shows OK:
3012/3012(Unspecified) (D) 255.255.255.255 0
We use complete regional blocks from Wizcraft and blocking at minimum
all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We
block almost anything that is not our actual customer market and screw
the rest.
On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavez cur...@telecomabmex.com wrote:
j...@sunfone.com wrote:
On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
yes I did that, even then i am not able to make outbound and inbound as
well.
That's weird. Guess you're gonna have to place a detailed ticket to
them. It sounds like a network problem to me but without any
On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
I am unable to register vitelity SIP trunk, where its keep on sending
registration request, and I am using Asterisk 1.4.39.2, my registration
procedure as follows,
sip.conf
register =
they were using pre-paid lines that ran out
money but they eventually got around and solved it. So I think that if
you insist with their support they usually resolve the issue.
Best,
--
Alejandro Imass
Cheers,
Jeff LaCoursiere
SunFone
On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander wrote
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote:
[...]
Just wanted to point out that after experiences with dozens of
termination
On Sat, May 12, 2012 at 5:05 PM, Eliezer Croitoru elie...@ngtech.co.il wrote:
On 10/05/2012 11:49, Bart Coninckx wrote:
Hi all,
for smaller (or maybe even bigger) sites I'm looking for a smaller,
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run
On Sat, Apr 14, 2012 at 7:55 PM, Joseph syscon...@gmail.com wrote:
I forgot to add:
clinic-amd*CLI iax2 show peers
Name/Username Host Mask Port Status
home_server (null) (D) 255.255.255.255 0
Unmonitored
iaxy-322/iaxy-3 (null)
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Yes, I have had no problems with Grandstream first gen ATAs, configured with
server and credentials and shipped off, they just work.
We use the HT-286, the server is on a public IP the nat setting on
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
[...]
The number of 'plain' SIP endpoints deployed behind consumer-grade NAT
devices talking to Asterisk servers on public IP addresses is in the
millions
argumentative line in this thread is that in our
particular case we found that IAX2 works great for _our_ set-ups and
we don't share the view that IAX2 is a broken bat, and that in fact
for us it just works great.
Thanks,
--
Alejandro Imass
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote:
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the discussion on this thread was that SIP is so awesome and
that IAX is a peice
use both SIP and IAX2
targeted to simple Home, SOHO and SMBs that just want to plug it and
work. We get that with IAX2 and not with SIP so from our experience is
completely the opposite of what you say.
--
Alejandro Imass
IAX2 is supported on cheap ATAs by several chineese companies and they
work
and off the shelf cheap ATAs,
period.
Also, respect netiquette and don't top post and use derogatory remarks
and keep your discussion technical.
--
Alejandro Imass
On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro
stot
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.com wrote:
On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote:
works. This cannot be done with SIP and off the shelf cheap ATAs,
period.
We do it, so cannot seems to be a strong word. It's not perfect,
Please
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote:
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote:
Please expand as to how you set-up a SIP ATA behind a common home
router set-up, without port redirection and/or use of a SIP proxy
and/or STUN
On Fri, Feb 10, 2012 at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking install some asterisk servers in a machine dell xeon
64 processor, but I'm not sure, about virtual Server
on extension 100
but I can't understand how because I don't even have an extension 100
declared anywhere. I would like to know how to block this MF because
he makes calls at 1-2 AM
--
Alejandro Imass
--
_
-- Bandwidth
too have fail2ban and running a relatively updated version of
FreeBSD. BTW my install is plain Asterisk
--
Alejandro Imass
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
, you could use virtualization or FreeBSD Jails for
example. Dunno how the telephony hw works with virtualization or jails
(yet, thoug I do have a single Asterisk running on a FBSD jail).
Good luck,
Alejandro Imass
Dave
On Sun, Nov 21, 2010 at 8:14 PM, Daniel Bareiro daniel-lis...@gmx.net wrote:
Hi all!
A few days I have problems connecting to the Internet on my house and
since then my local SIP extensions are no longer registered against the
local Asterisk server.
You have to be a bit more specific. For
this problem to my set-up or a quirky NIC but maybe it's
related to your problem (although it has _never_ happened to us in the
LAN extensions). Unable to find a solution, and since it's really very
rare, we have test calls every day to make sure everything is working
;-)
Best,
Alejandro Imass
On Sat, Nov 20, 2010 at 5:31 AM, Benoit Chabrier c...@chab.info wrote:
Thanks for your help.
you were right it also work without a stun server adding to sip.conf:
externip=78.47.x.x ; in [general] the IP of the dedicated server
nat=yes ; in the description of my peer
Exactly. BTW, IAX
can think of are NAT issues with SIP. If you are in
fact NATing try the Siemens phone to a direct IP to the server (no
NAT, firewall, etc.) and see.
--
Alejandro Imass
--
_
-- Bandwidth and Colocation Provided by http://www.api
do away with the stun server config in
the phone. Here is a great article that explains in detail the issues
with SIP and NAT: http://www.voipuser.org/forum_topic_7295.html
2010/11/19, Alejandro Imass a...@p2ee.org:
On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote:
Hello
On Wed, Nov 17, 2010 at 12:06 PM, Andrew Latham lath...@gmail.com wrote:
2010/11/17 Sevana Oy sa...@sevana.fi:
Hi,
Sorry for maybe not a very list related topic, but I have always been
curious if there is information on how many Asterisk based PBXs are
operating Worldwide?
Thanks and hope
On Wed, Nov 17, 2010 at 12:09 PM, Alejandro Imass a...@p2ee.org wrote:
On Wed, Nov 17, 2010 at 12:06 PM, Andrew Latham lath...@gmail.com wrote:
2010/11/17 Sevana Oy sa...@sevana.fi:
Hi,
Sorry for maybe not a very list related topic, but I have always been
curious if there is information
On Wed, Nov 17, 2010 at 4:04 PM, Andrew Latham lath...@gmail.com wrote:
Wouldn't apply to you, Steve, but sooner or later somebody will probably
imbed an innocuous phone-home into one of the Asterisk modules and it will
take a C person like yourself to point out the Microsoft-ness of this
-
|
|Asterisk
|
FAX -- FXS -
I'm using Asterisk 1.4.26.2 on FreeBSD 8.0
TIA,
Alejandro Imass
--
_
-- Bandwidth and Colocation Provided
On Mon, Aug 2, 2010 at 3:03 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Imass
Subject: [asterisk-users] FAX Options
[...]
TIA,
Alejandro Imass
IMO, as long as you're
On Mon, Aug 2, 2010 at 7:26 PM, Mark Scholten m...@streamservice.nl wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Monday, August 02, 2010 9:00 PM
To: Asterisk Users Mailing
into this.
Has anyone found a way to make pure Internet user-to-user Skype/SIP calls
via Asterisk (no PSTN involved) for free?
As I said above, once you have purchased your SIP channel you can make
free calls to your PBX using the special number but it's only INBOUND
AFAIK.
Best,
Alejandro Imass
. Finally, I resorted to try with
Asterisk 1.4 and it compiled fine. Conclusion: 1.6 won't work with
FBSD 7.0
Best,
Alejandro Imass
Thanks,
--Warren Selby
On May 25, 2010, at 9:48 PM, Warren Selby wcse...@selbytech.com wrote:
I was at a client site tonight to install OSLEC on his machine running
] WARNING[1790]: app_voicemail.c:7236 vm_authenticate:
Couldn't read username
Thanks beforehand!
Alejandro Imass
sip.conf
[101]
username=101
type=friend
secret=xx
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=home
mailbox=...@home
dtmfmode=rfc2833
extensions.conf
[home]
...snip
at all.
If it doesn't, somewhere DTMF settings need to be adjusted.
The IVR works fine, and we use it everyday. That's why it seemed to me
that it could not be a stmf problem. Any other ideas?
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-03-31 9:15 AM, Alejandro Imass
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
The message Couldn't read user name means it is not receiving the DTMF. Do
you have an IVR to verify that your system is receiving the DTMF? If not,
setup one, call into it and send Dtmf to it and see if it responds at
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