On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass wrote:
> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass 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) a
On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass wrote:
> 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
> th
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 OK
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 wrote:
> Yesterday a cus
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
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 detailed
info it's hard
gt; On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere wrote:
>>
>>
>> On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote:
>> > On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere
>> > wrote:
>> > > On Wed, 2012-05-23 at
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere wrote:
> On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
>> On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere wrote:
>> >
[...]
> Just wanted to point out that after experiences with dozens of
> termi
a and it seems 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, 2
On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N
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 => username:sec...@sip41.vitelity.net
On Sat, May 12, 2012 at 5:05 PM, Eliezer Croitoru 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 Asterisk fo
On Sat, Apr 14, 2012 at 7:55 PM, Joseph 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) (D) 25
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford wrote:
> On 2012-02-29 15:25:49 +0000, 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
way, my whole 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 jus
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming 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
> mil
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
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
asterisk is set to yes and wi
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez wrote:
> On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass 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 serve
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez wrote:
> On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass 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 p
SIP 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 wrote:
>> On Tue, Feb 28, 2012 at 6:36 PM, Steve Totar
think it works pretty well, and we 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
On Fri, Feb 10, 2012 at 9:18 PM, Carlos Rojas 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 software.
>
I use
o 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 for a li
sip vicious 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
--
_
, 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 FreeBSD 8. I have always
attributed 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
On Sun, Nov 21, 2010 at 8:14 PM, Daniel Bareiro 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 example is your A
On Sat, Nov 20, 2010 at 5:31 AM, Benoit Chabrier 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 doesn't ha
s 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 :
>> On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier wrote:
>>> Hello,
except sip.conf and
> extensions.conf).
> If you have any idea, please share it with me, i really don't to do to
> fix this problem...
> Thanks in advance !
The only thing I can think of are NAT issues with SIP. If you are in
fact NATing try the Siemens phone to a direct IP to the
On Wed, Nov 17, 2010 at 4:04 PM, Andrew Latham 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
>> snippet.
>
On Wed, Nov 17, 2010 at 12:09 PM, Alejandro Imass wrote:
> On Wed, Nov 17, 2010 at 12:06 PM, Andrew Latham wrote:
>> 2010/11/17 Sevana Oy :
>>> Hi,
>>>
>>> Sorry for maybe not a very list related topic, but I have always been
>>> curious if there i
On Wed, Nov 17, 2010 at 12:06 PM, Andrew Latham wrote:
> 2010/11/17 Sevana Oy :
>> 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 the community will
On Mon, Aug 2, 2010 at 7:26 PM, Mark Scholten 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
>
On Mon, Aug 2, 2010 at 3:03 PM, Danny Nicholas 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
t; FXO -
|
|Asterisk
|
FAX --> FXS -
I'm using Asterisk 1.4.26.2 on FreeBSD 8.0
TIA,
Alejandro Imass
--
_
-- Bandwidth
aid 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
>
> Thanks,
>
> Vieri
>
>
>
>
>
> --
> _
ever existed. 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 wrote:
>
> I was at a client site tonight to install OSLEC
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria 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 all.
> If it do
responds 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
0 10:30:23] 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
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