Hey folks,
I'm looking to potentially take some of my Asterisk servers and see
how well they fare in a cloud computing environment such as Amazon EC2
+ S3. I was curious to hear feedback from anyone who's willing to
share their experience if they've already done the same. Have you had
a positive
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage. This could be useful when you do
not have a backup T1
:
On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage
if they have failover though...
but they have iax2 and sip.
http://connect.voicepulse.com/ is their asterisk page.
Fred Posner
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
FWD#: 902963
On Jun 26, 2008, at 5:56 PM, Steve Finkelstein wrote:
We're personally
Hi all,
Would anyone be able to point me in the right direction as far as the
pros/cons of using a local loopback with a T1 provider, or just
peering with a company using SIP/IAX2 or my small office asterisk
setup? I've seen setups in both scenarios. The only potential pro of
the T1 that I can
I seen last time our providers rates.
Senad
On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
I have a budget to work with and was wondering if there are any
folks providing SIP/IAX2 trunking for unlimited inbound
...
but but for about $900 per month one could get T1 (24 channels)
unlimited in/out as far I seen last time our providers rates.
Senad
On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
I have a budget to work
Hi all,
I have a budget to work with and was wondering if there are any folks
providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate?
We're in the budget range of roughly $5,000 a month and we need multiple
channels per DID.
I'm not sure if something like this is feasible in
Hi folks,
I was recently made aware that the only way to currently set custom fields
in a relational database for CDR is via the experimental cdr_adaptive_odbc
drivers found here:
http://svncommunity.digium.com/view/tilghman/branches/1.4/cdr_adaptive_odbc.c?view=log
I had no problem compiling
: undefined symbol: ast_verb
[Dec 25 15:11:14] WARNING[22722]: loader.c:649 load_resource: Module
'cdr_adaptive_odbc.so' could not be loaded.
Thanks and happy holidays!
On 12/25/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 December 2007 12:50:06 Steve Finkelstein wrote:
I
(first use in this
function)
make: *** [cdr_adaptive_odbc.o] Error 1
On 12/25/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 December 2007 14:11:53 Steve Finkelstein wrote:
Any ideas on how to properly get this linked against my current
Asterisk?
catalyst*CLI module load
Thanks, that did the trick! :-)
On 12/25/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 December 2007 14:59:12 Steve Finkelstein wrote:
No worries!
Try updating from SVN now. I've made the change.
--
Tilghman
___
--Bandwidth
indecent when they let us down service. The actual VoIP
service is excellent; billing and paperwork can be messy at times.
Luki
On Dec 15, 2007 4:25 PM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
There's a myriad of options these days and I haven't been keeping up to
date
Hi all,
There's a myriad of options these days and I haven't been keeping up to date
with what's respectable any longer.
I essentially need a provider that will provide me with one DID to start and
let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on
my end and have full
Hi all,
I have the following in my extensions.conf:
exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 |
8585970327]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other CALLERID, it still directs me to s,15 which is the
Hangup() application. Here are
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a
for adding/removing numbers.
- sf
C F wrote:
It fails because the right function is ${CALLERID(num)}
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do
This might be of some assistance:
http://www.voip-info.org/wiki/view/vim+syntax+highlighting
- sf
Olivier wrote:
Hi,
New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor
with which I could easily edit Asterisk config files.
It seems Kate provide this type of service but
Using a quad core 1.86GHz Xeon CPU here, running Asterisk 1.4.2. Noticed
the following:
Cpu(s): 4.3% us, 95.4% sy, 0.0% ni, 0.2% id, 0.0% wa, 0.0% hi, 0.0% si
30908 asterisk 18 0 188m 10m 5152 S 400 0.3 51051:13 asterisk
Asterisk is eating up all the cores running the CPU at
What logic are you using to determine if the caller is indeed a
telemarketer, anyway?
Lacy Moore - Aspendora wrote:
I like to forward them back to themselves, that is, the ones that give
their phone number. Check nerdvittles.com. I think he had some kind
of torture script setup, if I
: [asterisk-users] asterisk telemarketer torture sound files
Manually,
I'm just designating some people as telemarketers (using some MySQL +
AGI to evaluate and drop the call) I would like to have a little fun.
- Adam
On May 6, 2007, at 12:06 PM, Steve Finkelstein wrote:
What logic
Unless there is some native rand() function available in Asterisk, I'd
look into writing a simple AGI using Perl, PHP or Python to return back
a random file to Playback().
More information here: http://www.voip-info.org/wiki-Asterisk+AGI
HTH
- sf
Jay Austad wrote:
I've got a directory under
With all due respect, I believe you might be a bit paranoid.
10-11M is quite normal for the linux kernel to allocate for asterisk.
It's not necessarily what the process is using, but that's just how
memory management works within the kernel.
What's 10-11M of RAM these days anyway?
- sf
Adam
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a Wait() within a Dial()
would
)
It waits 15 secs for the call to be answered
Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more
informations
Regards
Edoardo
Steve Finkelstein ha scritto:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other
Hi all,
This is a simple concept, however I'm not entirely comfortable with
available applications and functions available to me to make this happen.
I have a simple dialout macro such as the following:
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten =
Howdy Noah,
I just re-read my original inquiry and noticed my original purpose for
mailing the list was not simple to dig out of the message.
Ultimately, the dialout macro works fabulous. My issue is that I'd like
to be able to override one particular SIP endpoint with its own unique
callerID
If you are using a cisco switch (2950, 3560, CE500, 4000, 6500, or 3750)
then you will be able to setup the phone and have the computer daisy
chained to it.
I have a similar setup on mine. Here's how I configure my switch ports
in order to achieve the desired effect:
switchport access vlan 5
Interesting, that works David.
I got the example directly out of the published VoIP Hacks book and
followed instructions step by step.
Either way, thanks much. :-)
- sf
Dave Miller wrote:
Steve Finkelstein wrote on 4/28/07 12:21 AM:
my musiconhold.conf:
[default]
mode=quietmp3
Hi all,
I've compiled zaptel drivers and reconfigure asterisk afterwards from
source --with-zaptel.
Modules are loaded accordingly:
asterisk-1.4.2 # lsmod |grep z
Module Size Used by
ztdummy 5472 0
zaptel194504 5 ztdummy
crc_ccitt
I'd definitely purchase this text, especially if Theodore Wallingford
has any input on it. :-)
- sf
Doug Garstang wrote:
What a cool idea!
J. Oquendo wrote:
http://etel.wiki.oreilly.com/wiki/index.php/Main_Page
All,
I'm trying to hear the asterisk's auto attendant in its default
configuration. According to VoIP Hacks in Chapter 4, I found the
following excerpt after successfully configuring my SIP IP Phone (Cisco
7960G):
In its default configuration, Asterisk has an auto-attendant that can
route calls.
Hi all,
This is slightly off-topic, but I was hoping to be able to receive some
insight as I'm sure plenty of experts with c7960's exist on this mailing
list.
I'm attempting to upgrade from SIP 8.3 - 8.6 on a C7960G that I
inherited. I have my TFTP setup and unfiltered. The phone is doing TFTP
Hi all,
I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my
existing Cisco 7960G handset(s). I've tried multiple installs of
asterisk 1.4.2 with multiple handsets and SIP will not authorize my
phone. I'll include some verbose log messages below to show a VALID
registration and
Doug,
Were you also having issues specifically related to SIP authorization,
same as we're experiencing?
I noticed other folks on the list aren't having any issues either. Did
it just work out of the box?
Thanks ..
- sf
Doug Lytle wrote:
Steve Finkelstein wrote:
Hi all,
I've recently
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