of
magnitude.
It's just simple fact. Around 90% of fraud attempts against our network
come from that list.
And people, please stop trying to use human security to IP port analogies,
they make you look foolish.
--
Carlos Alvarez
TelEvolve
602-889-3003
chasing IPs for RTP, you are the one doing it
wrong. Is it SIP or RTP that's the problem in the firewall?
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New
. I don't think anyone here uses those.
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For any of you doing hosted PBX service on Asterisk, do you have a reliable
and secure click to dial solution? Particularly for Outlook, but since
about 20% of our customers use Mac OS, I'd love to hear about some that
work on that too.
--
Carlos Alvarez
TelEvolve
602-889-3003
a great
security model but one has to have their provisioning server for it to
function.
We've never had customers ask for this, but if doing so is fairly easy we
would look at it as just another feature we push. Do let me know how it
works out for you.
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Carlos Alvarez
TelEvolve
602-889-3003
? We currently use the Panasonic DECT
phones but they are a nightmare to configure.
If anyone wants to get in touch with them, our Grandstream contact is
Dennis Ryan, dr...@grandstream.com .
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Carlos Alvarez
TelEvolve
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connections
and call quality is 100% good.
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TelEvolve
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stuff in there where we do line counting and such.
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the customers manage their changes would also require some
interface you develop, and that part can get very complex because of things
like dialplan reloading. We do not allow client access to our hosted
PBX/IVR systems, so I can't advise you on that.
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Carlos Alvarez
TelEvolve
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and a
multi-tenant PBX. The IVR would just be in tenant contexts just like all
other features.
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that. I mean, not in one context
anyway, where the line numbers matter.
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group=Phone/Line_Key_2210/Short_Name_2_
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deploy custom hardware for specific functions or customers who are
particularly large in some cases. We just need a good reason to. Like
they want to self-manage, or they make a lot of changes, need custom
integration with databases, etc.
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Carlos Alvarez
TelEvolve
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to hear from one who is in a similar business.
Are you getting use of ARA too in similar hosted PBX offerings?
On Mon, Jul 30, 2012 at 10:00 PM, Carlos Alvarez car...@televolve.comwrote:
On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote:
2. Multi-tenant PBX: Configuring
We tried realtime and decided it wasn't for us. Never got it into
production, so I can't say much about it.
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Carlos Alvarez
Sent from my Nexus 7
On Jul 30, 2012 10:25 PM, Leandro Dardini ldard...@gmail.com wrote:
ARA is an acronym for Asterisk Realtime Architecture and is a different
way
I'm currently trying to decide on which GUI-enabled version of Asterisk to
use for one particular installation, where we will need good telecommuter
support. We've made it so easy for people to work remotely that the
customer is downsizing their real estate and will have 90% remote workers
with
arrives, and goes into the configure mode?
Is it end user friendly, or complex? I still don't have a server with the
Digium phone module on it to test.
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be comfortable with the configs
for either.
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the sendEmail program, which is hugely useful
throughout our systems for many purposes.
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the Digium phones over a Polycom
every time, that's an easy choice.
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of effort wasted in trying to save $40/mo just on this
list has already exceeded that savings for many years. A business with
one line and putting great effort into saving a pittance really needs to
rethink its priorities.
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Carlos Alvarez
TelEvolve
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. But again, since doing that, it's been totally
hands-off.
I will add though that we also use Fax for Asterisk simply to receive and
turn faxes into PDF for some customers, and that is perfectly stable also.
--
Carlos Alvarez
TelEvolve
602-889-3003
...@lists.digium.com] *On Behalf Of *Carlos Alvarez
*Sent:* Tuesday, May 15, 2012 2:18 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk forward call
On Tue, May 15, 2012 at 2:00 PM, motty.cruz motty.c...@gmail.com wrote:
Any suggestions how
for Asterisk to know it is
ringing. Why don't you just forward it using the phone company's call
forwarding service?
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New
run Asterisk on a Linksys WRT router, so I guess that's about a
minimum.
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, they are dirt
cheap and you're still getting a very good machine and good service.
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.
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a public IP at all? With most
routers, NAT from a public IP to a private one for the server works just
fine. I've used Cisco, Juniper, and Sonicwall in front of an Asterisk
server, NAT IP, and SIP with no problems.
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Carlos Alvarez
TelEvolve
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should works?
AGAIN: phones in one site behind nat and PBX in another site also behind
nat
No problem. I do that all the time. That's what I said in my message.
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On Thu, Apr 26, 2012 at 2:07 PM, Danny Dias ing.diasda...@gmail.com wrote:
Does not work for me!
What router(s) are in use?
Did you disable any SIP/VoIP helpers or ALGs they may have?
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Carlos Alvarez
TelEvolve
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port on the switch. We *never* run
anything other than Asterisk on a production voice server.
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New to Asterisk
the commercial version with a GUI and will tell you that the
cost/benefit is very clear. Great tool, pretty cheap ($1k I think).
Responsive support.
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= 16027170050/8774613644,1,Goto(monkeys,s,1)
Then:
exten = 1602717005X/8774613644,1,Goto(monkeys,s,1)
Etc.
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Carlos Alvarez
TelEvolve
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New
. Our system simply couldn't function without it.
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(starnetworks#extensions,9520,1)
exten = 1480241,s,Goto(starnetworks#extensions,9766,1)
exten = _X.,s,Goto(starnetworks#extensions|${EXTEN:7}|1)
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'.
exten = EXTENSION,PRIORITY,COMMAND
Nope, it's 's' priority. See my subsequent message about it. Priority CAN
be an 's' and in certain situations MUST be an 's' to function.
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Carlos Alvarez
TelEvolve
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the calls, then
each customer has a series of 's' priority lines for their individual DID
numbers.
I use the 's' priority in some DNIS/CID-based call blocking here and there.
As far as versions, 1.2, 1.4, and 1.6.
--
Carlos Alvarez
TelEvolve
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On Tue, Apr 3, 2012 at 8:42 AM, Vik Killa vipki...@gmail.com wrote:
#disasterisk fail
#freeswitch win
I suppose if you're too stupid to use real Asterisk, that's one alternative.
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Carlos Alvarez
TelEvolve
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the list know what region/country you're in, as
you'll want to be as close (i.e. low latency) to your trunk provider as
possible.
Also the amount of traffic and number of DIDs, since some of us serve
specific size ranges.
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Carlos Alvarez
TelEvolve
602-889-3003
a
little searching and you'll find many companies who don't see Canada
any differently.
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are SIP aware AFAIK, especially not the WRT-54G.
You can:
a. Continue to tell us that what we are doing every day is impossible.
or...
b. Go set up a test in your lab with an Asterisk server, a cheap
router, and a SIP client and see.
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Carlos Alvarez
TelEvolve
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, is generally trivial and
takes no special effort at all.
This is today's reality. That's why I mentioned the possibility that
he's using ancient devices and routers, or that Yugo.
--
Carlos Alvarez
TelEvolve
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system. This is
working fine, and we did it because of a strange bug/behavior with
certain versions of Cisco SPA series firmware.
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-alive packet is sent, the phone is no longer
reachable from Asterisk.
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particularly care.
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of the brand.
It's a piece of junk and everyone knows it, including the owners, so who cares?
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New to Asterisk
On Thu, Feb 23, 2012 at 2:21 PM, Faraj Khasib fkha...@iconnecths.com wrote:
Hi guys,
I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk
1.6
but I cannt make it work, can any body help me plz?
Errors and other details might be helpful.
--
Carlos Alvarez
TelEvolve
formulate a question that
gives you a useful answer.
Maybe we can start with a more basic question centered around exactly
what you are trying to do and why? Is the reason for asking because
you want to know how many ports to allocate in a NAT router, for
example?
--
Carlos Alvarez
TelEvolve
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commonly transported over UDP, not TCP.
And probably doesn't really answer whatever it is that he really
needed to know or do, since how many ports doesn't seem like
anything that would be a useful thing to know. Call it SIP trivia.
--
Carlos Alvarez
TelEvolve
602-889-3003
. This releases the Receptionist from the
call and the other 2 parties are joined directly.
I was just about to post exactly that. This is how we've taught our
customers to do it, the feedback is positive.
--
Carlos Alvarez
TelEvolve
602-889-3003
run 10m to saturation and still get good voice quality--if
you did it right.
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might not be
able to get to Y if you use carrier Z?
On Thu, Feb 9, 2012 at 10:25 AM, Bryant Zimmerman brya...@zktech.comwrote:
--
*From*: Carlos Alvarez car...@televolve.com
*Sent*: Thursday, February 09, 2012 11:25 AM
*To*: Asterisk Users Mailing List - Non
the users. None of them could
tell the difference, as they were just using everyday office applications
where this doesn't matter.
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.
I never work with analog lines, but I do believe that anything less than a
PRI would have the blocking done at the telco end before you get it.
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way to really know.
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Thurs:
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.
** **
I want it to also be able to register to the same asterisk box so it can
take calls and transfers.
What is the difference between a cordless phone and a wireless one?
We use and recommend the Panasonic KX-TGP500 and 550.
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Carlos Alvarez
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of totally useless wi-fi phones, they are all garbage.
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make them.
http://lmgtfy.com/?q=sip+phone+gigabit
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, however
that phone then goes into reorder tone.
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for the idea, I will ask.
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-train the receptionist...
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the oddities and have a suitable provisioning server set up,
deploying more is no problem. Troubleshooting is annoying because the
documentation is poor and there are lots of quirks/bugs/unexpected
features.
User acceptance on the Panasonic is very good.
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Carlos Alvarez
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service
provider like us where the configs travel the wild internet.
How do you like the Yealink phones? Is the cheap price worth it? We
looked at them a long time ago but there were some issues that prevented us
using them (centered around secure config over the internet).
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:
http://lists.digium.com/mailman/listinfo/asterisk-users
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There was an interesting flamewar one day in the Asterisk IRC channel over
Polycom love/hate. We fall into the hate category here, and hope to never
have to deal with them. If there was an SPA-series conference phone, we'd
all rejoice.
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Carlos Alvarez
TelEvolve
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peers
something that isn't going to be in the standard attack scripts. We use
letters.numbers format to name them.
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in a colo facility that are 7
years old and haven't been touched in at least two years. Just how many
more years do you need to be more stable?
Asterisk isn't perfect, but done right it's quite stable. IMO, the people
giving you advice just don't know how to do it right.
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Carlos Alvarez
TelEvolve
not
show up during normal testing. Now we only update for serious security
issue, and even then we try to handle the issue some other way.
Same here.
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without it?
Yes. A low-end single x86 server can easily support hundreds of endpoints
and dozens of concurrent calls, with all Asterisk services running on a
single server.
**
Do you have Asterisk expertise already? RoR, SQL, other telephony...?
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Carlos Alvarez
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! And again, thanks for your answer.
** **
** **
*Fra:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *På vegne af *Carlos Alvarez
*Sendt:* 14. december 2011 16:13
*Til:* Asterisk Users Mailing List - Non-Commercial Discussion
*Emne:* Re
and expect it to work. Your solid
advice might help me make the case for a fresh start, but whichever way it
goes, at least I’ll be kept busy ...
** **
*Fra:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *På vegne af *Carlos Alvarez
*Sendt
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A T1 cable according to this spec:
http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml
Crossing the 1/2 to 4/5 if needed.
On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr wrote:
2011/12/8, Carlos Alvarez car...@televolve.com:
I am
whatever I have around, which is typically Cat-5e. Yes, I know that
solid conductors aren't meant to be pushed into those connectors, yet my
experience is 100% good doing that.
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.
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.
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introductory webinar every Thurs:
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options visit:
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-alive fixes the problem. This is with Cisco/Linksys SPA phones, both
older 9xx series and the newer 303 and 5xx series.
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