Off topic but cool tech. One of the defunct Asterisk Appliance's was
based on the Altix 350.
http://www.sgi.com/products/servers/altix/numalink.html
On Wed, Aug 10, 2011 at 12:12 AM, Steve Totaro
wrote:
> I do love this stuff. Being self taught, Asterisk actually made me
> learn LAMP from the
I do love this stuff. Being self taught, Asterisk actually made me
learn LAMP from the command line. I was a pure Windows and Cisco guy,
all self taught since the VIC20 and Commodore 64, I taught myself
BASIC.
Anyways, just like anything you need to define the job before you
attempt to choose th
Guys in my opinion this thread has been very productive. Which proves
one thing, as many people you are going to ask about faxing with
asterisk that many opinions you are going to get (maybe even add +1
opinions :P).
In the end it depends on your experience, hence I asked the OP to try
for himself.
On Tue, Aug 9, 2011 at 6:00 PM, Steve Totaro
wrote:
> On Tue, Aug 9, 2011 at 5:21 PM, C F wrote:
>> On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan wrote:
>>> Hi,
>>> I would like to make sure I got it right:
>>> 1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
>>> fax from
On Tue, Aug 9, 2011 at 8:31 PM, Lee Howard wrote:
> Steve Totaro wrote:
>>
>> On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard
>> wrote:
>>
>>>
>>> Ryan McGuire wrote:
>>>
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38
Steve Totaro wrote:
On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard wrote:
Ryan McGuire wrote:
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38. The cheapest and
easiest approach that I have found is to buy yourself an FXS
On Tue, Aug 9, 2011 at 8:13 PM, Steve Totaro
wrote:
> On Tue, Aug 9, 2011 at 6:15 PM, Ryan McGuire wrote:
>> Unless your network is under load and you are seeing dropped packets
>> and high jitter, I would absolutely not do T.38. The cheapest and
>> easiest approach that I have found is to buy yo
On Tue, Aug 9, 2011 at 6:15 PM, Ryan McGuire wrote:
> Unless your network is under load and you are seeing dropped packets
> and high jitter, I would absolutely not do T.38. The cheapest and
> easiest approach that I have found is to buy yourself an FXS gateway
> and just make sure you are using u
On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard wrote:
> Ryan McGuire wrote:
>>
>> Unless your network is under load and you are seeing dropped packets
>> and high jitter, I would absolutely not do T.38. The cheapest and
>> easiest approach that I have found is to buy yourself an FXS gateway
>> and jus
On Tue, Aug 9, 2011 at 6:35 PM, Sassy Natan wrote:
> Ok,
> Thanks
> But what is the NVFAX? Does 1.4 support getting faxes per extension?
> In my office I have 1000 ext, each users has it's own DID number.
DIDs are cheap, get 1000 extra DIDs specific to that user's fax. It
is classy to have your
Ryan McGuire wrote:
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38. The cheapest and
easiest approach that I have found is to buy yourself an FXS gateway
and just make sure you are using ulaw.
As SIP is usually running over U
On 08/10/2011 12:00 AM, Steve Totaro wrote:
Try iaxmodem and hylafax. Alex B did a very nice writeup on how to
set that up so that it works very well.
Could not agree more. I have been using a box with an Eicon Diva Server
card, asterisk 1.4, chan_capi, iaxmodem & hylafax for many years and
Ok,
Thanks
But what is the NVFAX? Does 1.4 support getting faxes per extension?
In my office I have 1000 ext, each users has it's own DID number.
What I would like is that each user can get a fax using his own number.
I'm running version 1.4.29 under Debian, and FreePBX under a sip trunk.
It w
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38. The cheapest and
easiest approach that I have found is to buy yourself an FXS gateway
and just make sure you are using ulaw.
Is this for personal use, or for business use? One feat
On Tue, Aug 9, 2011 at 5:21 PM, C F wrote:
> On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan wrote:
>> Hi,
>> I would like to make sure I got it right:
>> 1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
>> fax from the PSTN and have anther FAX machine answer to it even if i
On Tue, Aug 9, 2011 at 3:38 PM, Sassy Natan wrote:
> Hi,
> I would like to make sure I got it right:
> 1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
> fax from the PSTN and have anther FAX machine answer to it even if it is
> behind asterisk. This works like any regula
http://www.101ftb.com/K00W10P513
Juan.
Linux User #441131
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.a
Hi,
I would like to make sure I got it right:
1. Asterisk 1.4 doesn't support FAX support. It do however works if u sent
fax from the PSTN and have anther FAX machine answer to it even if it is
behind asterisk. This works like any regular phone, and as far as I know
this mode known as T.38 pass t
Anyone?
On 11-08-08 09:37 AM, J Gao wrote:
Hello, All,
I have a question about using SRV record. One of SIP provider is using
DNS SRV record. If I use IP address of the SIP proxy server I can
successfully register my Asterisk 1.8.5. But If I try to use the
domain name like:
/register => use
That would work as well! :)
---
Marcelo Ellmann
Freeddom Tecnologia e Serviços S/A
+55 11 52133200 Ramal 1016
- Original Message -
From: "A J Stiles"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, 9 August, 2011 12:21:09 PM
Subject: Re: [asterisk-user
On Tuesday 09 Aug 2011, salaheddine elharit wrote:
> hello
>
> i want to know how to do in order to block all numbers bgin by 00 and all
> numebrs begin by 1
>
> i use sip account
All you have to do is, just make sure that there is no extension in the
default context which matches _00. or _1. --
I haven't used this function that much, but I'm pretty sure this will work out
for you
exten => s,n,GotoIF( $[ "${REGEX("[06]*" ${VARIABLE})}" != "1" ]
?INVALID_EXTENSION,1)
That should do the work. Spend some time reading the function documentation and
eventually you should get it right.
Thi
Thanks for your response
i have asterisk 1.4 installed , i configure some sip accounts 222 223 and i
want to block the outbound calls numbers begins by 00 an 1 because i want to
call just the local numbers in my country (all numbers in my country begins
by 06)
Could you please give me an exam
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Jim Boykin
> Sent: Tuesday, August 09, 2011 12:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DAHDI Ca
You could be a little more specific about your question but...
http://www.voip-info.org/wiki/view/Asterisk+func+regex
Just use the regex function. Any number which matches 00* || 1* goes to
invalid/block/whatever extension.
---
Marcelo Ellmann
Freeddom Tecnologia e Serviços S/A
+55 11 521332
hello
i want to know how to do in order to block all numbers bgin by 00 and all
numebrs begin by 1
i use sip account
thanks and regards
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New to Asterisk?
On Mon, Aug 8, 2011 at 3:57 PM, CDR wrote:
> I encourage the developers to check this out
> http://forums.asterisk.org/viewtopic.php?f=1&t=77692&p=161590#p161590
>
> I am calling from behind a NAT, and there is no way to force Asterisk
> to stay in the path. If the codec is the same as the outbou
37pgn.
http://darkskiesblog.com/wp-content/uploads/img/vosc.html
22gv7con3pr fjfvojvm e1cugvfj, tuzj2tcz2 4zssju6k5bfj. jdc52cc
twdoh35sn4s sb2yj.
--
Thanks,
Max Alex
Voip Developer
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Update:
Yesterday I did not observe any unexpected traffic.
So far so good.
Thx
Sans
On Mon, Aug 8, 2011 at 9:24 PM, Антон Квашёнкин wrote:
> Ok, run your script and then do this:
> service iptables save
>
> And by the way, list "chkconfig --list iptables" output.
>
> 2011/8/8 RSCL Mumbai
>
On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote:
> Hi
>
> I'm using asterisk 1.8.3.2 (with a couple of patches)
>
> I have the following scenario...
>
> SIP call comes in and gets answered by extension A (MixMonitor is
> executed as part of this inbound dial plan of the number being called
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