On Thu, 2011-02-17 at 11:12 -0600, Danny Nicholas wrote:
...Please help me as of i am doing my master thesis on evaluating the
performance of various open source projects
Thanks in advance
Awaiting for the reply as soon as possible,
Awesome. Any institution that issues a
On Tue, 2011-01-18 at 15:18 +, Andrew Thomas wrote:
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?
Top posting is here - to stay!
Stop
On Mon, 2011-01-17 at 02:31 +, James Miller wrote:
I hate to disagree but I find it much, much easier to follow conversations
when the newest reply is on top. I find it too time consuming to scroll
through a long message just to find out someone left a three word reply.
As I am on my
on, Always Connected
-Original Message-
From: Fred Posner f...@teamforrest.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 16 Jan 2011 21:43:00
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing
On Dec 9, 2010, at 5:57 AM, Joe Greco wrote:
Hello,
We had been seeing SIP-guessing attacks on our Asterisk server here.
While it wasn't that hard to write a once-a-minute cron job to spank
the lusers, that runs once a minute and creates little spikes in the
usage and I/O graphs,
On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote:
Hey, all. I'm in the middle of a rollout, and just learned that the
SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued.
The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130
more/handset. AND it
On Nov 5, 2010, at 10:45 PM, Michael Graves wrote:
On Fri, 5 Nov 2010 19:02:43 -0400, Mike wrote:
On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote:
Hey, all. I'm in the middle of a rollout, and just learned that the
SoundPoint IP 430 -- my favorite mid-range phone -- has been
On Nov 5, 2010, at 11:13 PM, Michael Graves wrote:
On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote:
Curious Michael... Why won't you subject people to the 335's? I love these
phones for a call center deployment. The are a fantastic agent phone... let
alone a great phone for kitchens
On Nov 4, 2010, at 9:41 AM, C F wrote:
You see the problem is that asterisk will send as many packets as its
admin does on the list. There is no way to change that. I suggest you
first change the amount of packets per second you send.
On Thu, Nov 4, 2010 at 5:38 AM, ali anjum
On Oct 21, 2010, at 11:11 AM, Steve Howes wrote:
On 21 Oct 2010, at 15:56, JR Richardson wrote:
These are full time positions in Dallas, no telecommuters please.
A very vast majority of people on here are not in Dallas (and indeed probably
a majority in the US). So stop filling their
=yes
---fred
Fred Posner
http://qxork.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org
I have a decent thread on Team Forrest about this. Also, good to read up on
John Todd's 7 deadly sins.
http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block/
---fred
http://qxork.com
On Oct 3, 2010, at 4:51 PM, Alec Davis wrote:
Make sure you have allowguest=no in
On Sep 27, 2010, at 1:29 PM, Michelle Dupuis wrote:
That's exactly what we recommend for DB/realtime installs. HAAST's focus is
the failover, promotion, assignment of IP, etc. but links to standard tools
for file/db sync. In line with the philosophy of try to not be everything to
I wrote a script to help with these here:
http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block
To each their own... there's 1000 ways of combatting this.
---fred
http://qxork.com
On Sep 17, 2010, at 5:18 PM, dave george wrote:
I am getting several hundred
On Jul 30, 2010, at 5:04 AM, Andraž wrote:
Ok, problem is another, when I run configure, it write this:
checking for tds_version in -ltds... no
configure: ***
configure: *** The FreeTDS installation on this system appears to be broken.
configure: *** Either correct the installation, or run
this in some reasonable way.
sean
What is the default context in sip.conf? Does it allow outbound calls?
Do you have autocreatepeer=no?
Fred Posner
http://qxork.com
--
_
-- Bandwidth and Colocation Provided by http://www.api
On Jun 11, 2010, at 8:03 PM, sean darcy wrote:
Fred Posner wrote:
On Jun 11, 2010, at 5:55 PM, sean darcy wrote:
snipped...
What is the default context in sip.conf? Does it allow outbound calls?
;###
;DEFAULT CONTEXT
;###
[default]
exten=_1XX,1
On May 20, 2010, at 12:43 PM, Myles Wakeham wrote:
I am trying to implement a change to our Dialplan that will thwart
tele-spammers that are calling us with blanked out caller ID.
The caller IDs seem to vary between originating callers when they block
caller ID. I've seen the following:
On May 18, 2010, at 1:13 PM, Don Kelly wrote:
Has anyone had good results with an on-line database that returns a LATA
based on NPA NXX?
--Don
Don Kelly
There's an online list that you can convert to a locally stored db.
http://www.nanpa.com/nanp1/allutlzd.zip
---fred
Same problem here.
---fred
On May 17, 2010, at 6:28 AM, Alexandru Oniciuc wrote:
kb.asipto.com isn't reachable: DNS doesn't resolve the domain name.
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per
On Apr 28, 2010, at 11:30 AM, Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at all
3 stages but with different parameters on the command line to indicate
the call status. Works fine before the call is
On Apr 28, 2010, at 1:00 PM, Gareth Blades wrote:
Steve Edwards wrote:
Steve Edwards wrote:
How do you reconcile your assumption that the Perl module is reading
STDIN and your statement that your AGI quits before asterisk has
finished sending the information about the current call via
On Apr 28, 2010, at 1:12 PM, Steve Edwards wrote:
On Wed, 28 Apr 2010, Fred Posner wrote:
Did I miss where the code was posted?
Yes. In my mail reader it is Gareth's second post.
Thanks. Wish I hadn't looked now.
--fred
http://qxork.com
On Apr 21, 2010, at 4:50 AM, Gordon Henderson wrote:
On Tue, 20 Apr 2010, Frank Bulk wrote:
Please take note of their posting:
https://aws.amazon.com/security/
which discusses the issue and what they're doing to improve response.
And is anyone on the list worthy of being considered
On Apr 20, 2010, at 6:18 PM, Frank Bulk wrote:
Please take note of their posting:
https://aws.amazon.com/security/
which discusses the issue and what they're doing to improve response.
Frank
If only they wrote the truth...
When we find misuse, we take action quickly and shut it
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sun, Apr 18, 2010 at 12:10:32PM +0200, Randy R wrote:
Hi,
We all know most people are reporting that Amazon hasn't been helpful
at all. A few people say they've received answers, but most are
getting smoke screen PR BS.
You can vote this
On Apr 18, 2010, at 1:14 PM, Randy R wrote:
On Sun, Apr 18, 2010 at 6:02 PM, Stuart Sheldon s...@actusa.net wrote:
For what it's worth, here is my Blog Article from the incident...
http://www.stuartsheldon.org/blog/2010/04/sip-brute-force-attack-originating-from-amazon-ec2-hosts/
Saw it
On Apr 13, 2010, at 8:04 AM, Hans Witvliet wrote:
On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote:
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out
On Apr 13, 2010, at 4:22 PM, Randy R wrote:
On Tue, Apr 13, 2010 at 8:25 PM, Steve Murphy m...@parsetree.com wrote:
Hmmm. It would seem that it would be to Amazon's advantage to jump on this
problem,
I am pushing for this, please everyone who is suffering from this
problem, submit it or
On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote:
Perhaps if there was a Asterisk RBL we could all contribute to; for which we
could then hook into and drop any connection where a source IP is listed ?
--
Thanks, Phil
I love the idea of a RBL... count me in for contributing.
On Apr 12, 2010, at 1:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project. It's a
blacklist of IP addresses and address ranges which are known to ONLY be
used for malicious
On Apr 12, 2010, at 4:50 PM, Chris Hastie wrote:
I'm currently receiving over 200 SIP REGISTER requests per second from a
machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it.
This has continued for several days, and ab...@staff.aruba.it are
unresponsive. I've had a couple
On Apr 11, 2010, at 10:06 AM, Zeeshan Zakaria wrote:
I don't k know if there is a tool to sniff passwords, but did you check in
/va/log/asterisk/full? Maybe wireshark can be used for this purpose, but
it'll be not that straight forward.
Interestingly I checked log of my server and found
On Apr 11, 2010, at 4:06 PM, Tom Stordy-Allison wrote:
Hi,
This is exactly what I've just joined this mailing list about.
Has anyone has any luck getting Amazon to stop the instances? I'm stuck with
around 700Kbps of my 2.5Mbps inbound in use as my firewall blocks the
requests as
On Mar 17, 2010, at 1:05 PM, Bruno Camargo wrote:
Hi Giorgio,
So it means that Asterisk has no native support for g729 ?
Thanks
--
BrCaBadT
--
Depends on your definition of support. It supports passthrough... but if you're
using it locally on a bridge on transcoding, you'll need
On Mar 8, 2010, at 6:16 PM, sean darcy wrote:
And without doing anything more, it now Just Works(TM). Sunspots possibly.
sean
Glad it's working... those sunspots are nasty. :)
---fred
http://qxork.com
--
_
-- Bandwidth
On Mar 5, 2010, at 1:01 PM, sean darcy wrote:
The issues are that sip doesn't work,
What does doesn't work mean? In / Out? Both? Do you have a sip trace?
even though this same set up
worked with POTS dsl. IAX does (but gives lousy audio quality) so I
don't believe all udp ports are
On Mar 3, 2010, at 1:03 PM, sean darcy wrote:
Well at least my RG doesn't let you use DMZplus _unless_ you've chosen
dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh
into my router from the internet.
Anybody else got this working?
sean
What are the issues?
On Mar 2, 2010, at 2:37 PM, jonas kellens wrote:
Does Asterisk know when it hits a voicemailbox ?
When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.
If Asterisk does not know it's a voicemailbox that has answered the call, the
voicemailbox
On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
I've just got Uverse installed. I had dsl, but ATT insisted I couldn't
keep my old dsl, but had to switch to Uverse internet - vdsl.
My setup:
linux box as router : 10.10.11.252
asterisk box: 10.10.11.180
10.10.11.252 is
On Jan 29, 2010, at 5:59 AM, Olle E. Johansson wrote:
29 jan 2010 kl. 10.25 skrev Alex Balashov:
I don't know about 4xx, but 503 would be more benign for general/
miscellaneous errors than 603.
503 indicates that there's a problem with the server, so that's not a good
replacement.
On Jan 16, 2010, at 1:16 PM, Mr. James W. Laferriere wrote: [snip]
Kevin , Sometimes your about as helpful as passing wind .
Hmmm... I do find passing wind to be quite helpful sometimes. Afraid I lost the
reference.
---fred
--
On Dec 28, 2009, at 8:03 PM, Taylor, Jonn wrote:
Darrick Hartman wrote:
-Original Message-
On Behalf Of Rick Huebner
Sent: Monday, December 28, 2009 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Looking at Asterisk for 8000sq/ft
On Dec 23, 2009, at 9:48 AM, Danny Nicholas wrote:
Why are there three branches of 1.6?
http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/
There's more info on the blogs about the new method and long term releases.
___
On Dec 23, 2009, at 4:21 PM, Sascha Ferley wrote:
Hi,
I am in need of ordering a new server here for our asterisk solution. Since
the corporate standard is Dell we need to stick to a dell server. We used to
deploy 2900III without any issues, however now they are almost not available
any
On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:
Dear All
I have an application that calls for my Asterisk sip to be connected to an
external sip server for voip routing . Please be informed that my Asterisk
sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this
end
On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote:
On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:
Dear All
I have an application that calls for my Asterisk sip to be connected to an
external sip
On Dec 17, 2009, at 10:36 PM, Neeraj Chand wrote:
Just finished with the instructions from digium website/ net on how to
compile FFA:
After restart, modules did not get loaded so tried to load manually:
[Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module:
Error loadin
On Dec 13, 2009, at 7:20 PM, Jerry Geis wrote:
I have been looking for a way from the dialplan to inquire if there are
any members in a queue.
So what I want to do is if no users are members of a queue then I can
send the call to a given extention.
I have the queue setup all that is
On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote:
Hello,
We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
like to use the sip,extensions and voicemail in realtime mode.
Where can we find the database tables structure for these versions?
Thanks,
Andreas
This is
On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?
Don't know of any wifi phone that works flawlessly whatsoever. Best to consider
a DECT style phone.
On Dec 3, 2009, at 5:05 PM, Matt Riddell wrote:
On 4/12/09 9:28 AM, Scott L. Lykens wrote:
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with high
availability without dropping calls.
See
On Nov 15, 2009, at 2:27 PM, aster...@opensourcesolution.in wrote:
i am going to set up asterisk for pbx purpose in my office. i am having 2
PSTN lines and will be configuring 10 extentions in my office. plz tell me
which hardware will be needed for this.
thx
Have you read this page?
On Nov 13, 2009, at 6:16 PM, Cary Fitch wrote:
My point was the two previous posters could have ignored the request and
made no post at all. That they were violating a rule by top posting to
tell a person not to bug them.
And, someone criticized me for an off topic post and of course
On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote:
I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
(1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
Server 9.04 with the default Debian package manager installation of
Asterisk. (version
On Sat, Nov 7, 2009 at 4:45 PM, John Timms johngti...@gmail.com wrote:
Hi Fred.
By fast I mean the best Business DSL Bellsouth has to offer: Up to
6.0 Mbps downstream - Up to 512 Kbps upstream
If you're running the GSM codec, 7 calls will hit around 200 Kbps. If
you're running ulaw, 7 calls
Zoa,
It's Michael Graves... www.mgraves.org
Sincerely,
Fred Posner
f...@teamforrest.com
+1.503.914.0999 (direct)
On the web at http://www.teamforrest.com
On Oct 19, 2009, at 11:58 AM, Zoa wrote:
I missed the talk that was given on wideband codecs @ astricon last
week.
I tried
On Tue, Oct 13, 2009 at 11:22 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Tue, 13 Oct 2009, David Wathen wrote:
... What firewalls work good with VOIP?
...I use Draytek Vigor 2820's these days. ...
Gordon
I've had great luck with Untangle. Open Source... can shape traffic
factor system and is truly great.
Fred Posner
f...@teamforrest.com
+1.503.914.0999 (direct)
On the web at http://www.teamforrest.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
configuration.
Fred Posner
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options
On Sep 3, 2009, at 2:34 AM, Olle E. Johansson wrote:
2 sep 2009 kl. 22.40 skrev Fred Posner:
Here's the story...
Nortel system set to use g711 @ 30ms payload ... Asterisk box would
need to communicate to that box @ 30 ms and another end point at 20
ms.
I've seen discussions of setting
.
Anyone have luck with this?
The Asterisk can be 1.4 or 1.6.x... I've a preference for 1.6.0.x but
it's not set in stone :)
Fred Posner
f...@teamforrest.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009
Anyone hear anything? Very down for me right now.
Fred Posner
f...@teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
their fault. In every other respect, I've
been
happy with their service.
I should note I'm a low-volume customer, residential and home office
use.
Paul
Also very happy with flowroute.
Fred Posner
f...@teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
On Jan 23, 2009, at 12:44 PM, Steve Edwards wrote:
On Fri, 23 Jan 2009, Steve Totaro wrote:
Also, can speed up complaints of a slow network...
Just what we all need -- faster complaints :)
TMC posted an article on the Packet8 DNS outage:
throughout.
I had been using Voicepulse, but suffered more downtime last year than
I hoped and found that support was a little hard to get a hold of when
issues occur.
Fred Posner
f...@teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
On Jan 13
of menu on * or 0, not using the o or a
variables.
Fred Posner
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
Starting around 10:00 AM EST.
All services from them whether I connect by IP or DNS (both east coast
and west). Anyone else?
Fred Posner
f...@teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
Yeah, they finally updated via their twitter account... Seems a
generated exploded.
http://www.voiptechchat.com/voip/165/voip-carrier-voicepulse-suffers-outages-uses-twitter/
Fred Posner
www.teamforrest.com
On Dec 22, 2008, at 11:15 AM, Shane Young wrote:
Quoting Fred Posner f
3 months of me buying it.
I have a few grandstream 286's I like to use for traveling and placing
in remote areas of an installation.
Fred Posner
___
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asterisk-users mailing list
All you need is odbc and freetds. Then it will integrate very smoothly.
Fred Posner
f...@teamforrest.com
Direct: +1 (503) 914-0999
-Original Message-
From: Steve Wofford s...@uctrlit.com
Date: Thu, 18 Dec 2008 19:46:36
To: Asterisk Users Mailing List - Non-Commercial
/2008-December/223172.html
--
Michiel van Baak
And for a rant, see this:
http://www.voiptechchat.com/voip/146/fbi-security-warnings-and-voip/
-Fred Posner
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
it have? Mine were v1.22 just the other day when I checked, but I
have it set to D/L the current beta automatically once a day.
Michael
--
Mine automatically updated as soon as it got an IP address. (default
setting)
Fred Posner
[EMAIL PROTECTED]
Main: +1 (212) 937-7844
Direct
Anyone know of where to get a Manilla or Philippines DID? I show (1)
on didx.net but is rated too low to purchase.
Fred Posner
[EMAIL PROTECTED]
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
On Wed, 5 Nov 2008, Pedram M wrote:
Any recommendations on good wireless SIP phones?
VoIP Tech Chat did a review on the Linksys WIP 330:
http://tinyurl.com/review330
and VoIP Supply has a new phone (haven't read any reviews) that has a
new long-life battery.
Fred Posner
smime.p7s
On Nov 3, 2008, at 12:11 PM, Robert Augustyn wrote:
Is there anything like that?
Any experiences?
X-Lite is a free download and has video capabilities.
Fred Posner
[EMAIL PROTECTED]
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
smime.p7s
Description
as well as a lack of spy/whispering commands
available
via Asterisk Manager. Does anyone know how to implement this?
Thanks a lot.
Regards,
Victor
If it's a pre-paid app and you're doing a Dial command (like a calling
card), why not use the limit (L) feature that's built in?
Fred Posner
options with meetme such as usercount, admin
mode, etc.
Good write up here:
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
Main: +1 (212) 937-7844
South: +1 (352) 379-7334
calls to a2billing context as follow:
sip.conf
[sip_proxy1]
type=peer
context=a2billing
host=81.201.82.39
dtmfmode=RFC2833
rfc2833compensate=yes
Try adding:
relaxdtmf=yes
to the peer
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
www.teamforrest.com
Using VoIP?
SIP:[EMAIL
On Sep 25, 2008, at 9:00 PM, Darrick Hartman wrote:
Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the
account
connected to the Asterisk server.
Lots of questions about this one. There's definitely a demand for
it so
I can see why Digium would be
, on the callin, you can have multiple channels. It's
really very exciting.
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
Using VoIP?
SIP:[EMAIL PROTECTED]
smime.p7s
Description: S/MIME cryptographic signature
On Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote:
Make host=dynamic.
Also, set nat=yes
Hi all,
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host= 192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
Fred
of the
NAT address. Also, add nat=yes if you're doing that.
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
www.teamforrest.com
Using VoIP?
SIP:[EMAIL PROTECTED]
smime.p7s
Description: S/MIME cryptographic signature
___
-- Bandwidth
Fred,
The context should stay friend or i should change it to another thing?
Regards
This would depend on what you want that user to be able to do...
Here's a good source to learn the differences:
http://www.voip-info.org/wiki/view/Asterisk+sip+type
Fred Posner
[EMAIL PROTECTED
On Aug 29, 2008, at 8:18 AM, Karl Fife wrote:
I've had essentially no problems with my snom m3s. Someone from snom
has been in touch to confirm that they are now putting more effort
into
the firmware for this phone. There are a few new features that I'd
like
to see that are already in
On Thu, 28 Aug 2008, Jaap Winius wrote:
Hi list,
Are there any reliable wireless SIP phones available on the market
yet?
My Linksys WIP330 should arrive today. I've always wanted to test how
well it would work in hotspots... will let you know.
Fred Posner
[EMAIL PROTECTED]
Tel: +1
Anyone else having timeouts to Voicepulse?
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
___
-- Bandwidth and Colocation Provided
On Thu, 28 Aug 2008, Jaap Winius wrote:
Hi list,
Are there any reliable wireless SIP phones available on the market
yet?
My Linksys WIP330 should arrive today. I've always wanted to test how
well it would work in hotspots... will let you know.
Fred Posner
[EMAIL PROTECTED]
Tel: +1
= start,n,Queue(ivr|tT|||30)
exten = start,n,voicemail(ACCOUNT)
Fred Posner
www.teamforrest.com
smime.p7s
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AstriCon 2008
http://tech.slashdot.org/article.pl?sid=08/08/19/2229245from=rss
Any comments?
Cheers,
Dean
I hate those calls and have been getting them more and more. It
sounds like there will be no enforcement until early next year, but
good riddance.
The worst one is a company that sells extended
Anyone know where I can get an incoming DID for Bogota, Colombia?
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
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dirty ;)
Fred Posner
[EMAIL PROTECTED]
www.teamforrest.com
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register
a success or error.
Honestly, if you're running a webservice, I like using the CURL
function. Works like a charm for me.
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
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On Jul 15, 2008, at 10:20 PM, Steve Edwards wrote:
curl() doesn't fire up another process. The response is returned as
just
one big chunk. In my case, it was the HTML to an entire web page :)
If you need to do a bunch of parsing, maybe an AGI calling libcurl --
saving a bunch of ugly
.
I use a blacklist script to add numbers to and record the call. During
that 5 sec wait they normally talk, so it provides me some sense of
amusement (sadly).
Fred Posner
[EMAIL PROTECTED]
www.teamforrest.com
FWD#: 902963
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,
especially next Friday July 4th.
tia,
Randy
I love it. I'm celebrating the 4th with a 2000 mile motorcycle ride :)
I'll do my best to make it for the conference.
Fred Posner
www.voiptechchat.com
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On Tue, Jul 1, 2008 at 7:30 PM, Fred Posner [EMAIL PROTECTED]
wrote:
a 2000 mile motorcycle ride :) I'll
Where to where?
Gainesville, FL to Ann Arbor, MI to Gainesville, FL
What Michael said.
I had a blueant bluetooth, which was awesome on the motorcycle. Clear
and dialed the number?
Usually best to use WaitExten() or TIMEOUT(digit). For most stuff I
do, I just use waitexten.
Fred Posner
[EMAIL PROTECTED]
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I think Voicepulse is out of NYC... not sure 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
Have you tried keeping asterisk in on the call with a /n connection in
the dial-plan?
Is there any firewall that is blocking udp ports to any of your clients?
Fred Posner
[EMAIL PROTECTED]
On Jun 21, 2008, at 12:36 AM, Sam Tam wrote:
Well to be honest, our experience with asterisk never
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