On Thu, Feb 25, 2010 at 8:00 PM, David Gibbons d...@videon-central.com wrote:
Duh! How are we going to spread the word about how to take those alien
bastards down if we don't keep morse code around!?!??!
And what about if you're trapped in ship that sinks? What if the 3g
coverage isn't good?
On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News) alansli...@gmail.com wrote:
Another vote for the Siemens Gigaset range. Been using the S685IP almost
since the day it was released here in the UK. Nice handsets, great voice
quality, but as others have said the UI can be a bit slow.
Alan, don't
On Tue, Feb 23, 2010 at 10:50 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
High quality to me means well built, reliable, good protocol support and
above all a responsive manufacturer.
Incidentally, I've dropped two of the S675IP handsets on the hardwood
floor a few times, still working fine.
On Tue, Feb 23, 2010 at 7:50 PM, Danny Nicholas da...@debsinc.com wrote:
What I want is, if a call coming from a trunk 100 rings, and if the
caller wants to be transfered to 101, the transfer is denied. In other
words, 101 can't get transfered calls.
WHat about using featuresmap to replace the
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote:
You propably have a type=friend where the user part matches before you even
hit the peer part, where the insecure configuration parameter matches. There
is a confusion here on the From: username and the authentication
Hi,
When Jason Goecke talks, VoIP ideas become reality, and this makes my
day. On this call we’ll talk about the newest features in Tropo and
how to get started with telephony apps in the cloud without adding new
infrastructure. Here's a chance to speak directly to Jason (or JSON as
we now call
On Mon, Feb 15, 2010 at 9:51 AM, Olle E. Johansson o...@edvina.net wrote:
To avoid extensive rewriting and fix the current issue.
That works in countries where you have fixed-length numbers. Unfortunately,
not every dialplan works that way, so that can't be a generic advice even
though it
Hi,
Barring WW (wifi woes), I will be broadcasting live from the HD
Communications Summit this Friday. Usually we begin at 12 Noon EST but
we may start earlier so please check the site, IRC, Twitter or
Facebook for the exact start time. If any of you are planning to be
there, please email me if
On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote:
What I have seen on my asterisk box when I had a up/down adsl line was
that the asterisk box couldn't do dns resolution and would hang( well no
other internal calls could be made, seemed like some sort of semaphore
was
Why not run a internal DNS with forwarders to your ISP ? That way Asterisk
can still resolve itself and hosts internally.
See above:
you need a local
resolver, like a caching BIND server, on the same host.
Nice, but still, it ruins the all in one concept. Isn't there a
lighter solution?
On Fri, Feb 5, 2010 at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Doh! :) My philosophy has always been to install a local named server,
whether it be for Asterisk or something else, as most of the time everything
I do is behind NAT and I prefer to resolve internal addresses. This
2010/2/5 Vinícius Fontes vinic...@canall.com.br:
Have you tried to set srvlookup=no on your sip.conf?
I think that just stops SRV lookups, not regular DNS.
/r
--
_
-- Bandwidth and Colocation Provided by
Hi all,
OT but possibly of interest to many of you in the asterisk community,
Markus Feilner is our guest tomorrow on the VUC: VPN Users Conference.
Markus is an interesting guy. In a former life, Markus ran an asterisk
box and used Sipgate.de. He works for a German Linux publication and
just
Hi,
In the aftermath of Digium's and Counterpath's Bria for Asterisk
announcement, we're happy to chat with Todd Carothers, Counterpath
Product Manager today at 1 PM EST.
For more info, http://vuc.me
Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc
Call in
I realize that many of you are too far away to consider it, but I know
of a couple of people who are considering going. Is anyone tempted? I
am planning on going and have a promo code for you if you'd like one.
r
--
_
--
The problem 'I can place calls but no one can reach me'
is our number one support question. Advising the user to check the DND
As a general comment, the DND button on a decent phone should LIGHT UP
when it's in use. On the Polycom 650, it is very clear on the LCD
screen with flashing icons, but
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote:
exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
but what should i do. if i want to set seperate weekdays,like mon,wed.
not continuous weekday like mon-fri.
I couldn't find any reference to multiple, non-contiguous
On Fri, Jan 22, 2010 at 1:26 PM, Julian Lyndon-Smith aster...@dotr.com wrote:
Anyone got any subjective (!) views on the merits of these two ranges ,
using asterisk 1.4 ?
The choice of phones is crucial. Setting aside my tastes, you really
need to get a couple of typical users to try them
http://twitpic.com/z8n36
On Fri, Jan 22, 2010 at 8:11 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hey hey!
Anyone got any subjective (!) views on the merits of these two ranges
, using asterisk 1.4 ? I need to supply approx 30 handsets to a new
client, with the
On Wed, Jan 20, 2010 at 4:40 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
The AstLinux Team would like to announce that the 0.7.0 version of
AstLinux is available for download. There have been many significant
updates in this release including updating to the latest Asterisk
Release
Hi,
Our guest this Friday is Himanshu Dwivendi, author of the book Hacking
VoIP. You're welcome to come discuss it with us on the conference.
Find your local time by going to http://vuc.me/next - the conference
begins a little before 12 Noon Eastern Time.
VUC has an IRC channel #vuc on
About what?
On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote:
On 8 Jan 2010, at 13:52, John Novack wrote:
Steve Howes wrote:
On 8 Jan 2010, at 02:28, John Novack wrote:
Careful, or Steve will un top post YOU!
I like it in the past. Leave me alone ;)
Different
Hello,
In about one hour we should be chatting with Tim Behrins of Voxbone
about their initiative, iNum. I say should because he's the
scheduled guest, but I haven't heard from him today :)
Next week, we'll be Hacking VoIP
Feel free to top post your answers, it seems to stimulate conversation.
On Fri, Jan 8, 2010 at 5:25 PM, David Gibbons d...@videon-central.com wrote:
I would have read your message but I couldn't find it amongst all of this
garbage...
Funny I saw your right away :)
Ok, all kidding aside, I really don't care where people post if only
they'd clip all the garbage
On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i want to dial a number to let two phone ring at the same time or
alternative ring,
how should i configure in asterisk? or how to right the Dialplan code?
exten = 12345,1,Dial(${PHONE1}${PHONE2})
each phone variable
On Thu, Jan 7, 2010 at 3:07 PM, Zhang Shukun bit...@gmail.com wrote:
Thank you!
but how can i determine whether ring at the same time or
alternative ring?
BTW, the uri
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
It got mistyped or cut, it's
On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones allan...@gmail.com wrote:
But jailbreaking increases the freedom to develop a application and
Oh, I agree with you, but it's probably even better to make a decision
to either buy into the constraints of Apple or find a better, free-er
phone, which is
On Tue, Jan 5, 2010 at 8:53 PM, UIT DEVELOPMENT uit...@gmail.com wrote:
I've been poking around the past few weeks, trying to familiarize
myself with all of this. I am new to Linux, VoIP and Asterisk -- to
be complete. This is my first exposure to all of these technologies.
I think one of
On Wed, Jan 6, 2010 at 6:48 AM, Allann Jones allan...@gmail.com wrote:
Jailbreak your iPhone and install Cydia to have a Unix like open
source environment (based on Debian), then install Siphon SIP client,
and have fun!
There are at least 4 iPhone SIP clients available for $3-10 that work
well
Thanks to Digium, the company, and to all of the fine people from
Digium who participate in the weekly VoIP Users Conference conference!
We will be live on Friday January 1, 2010 and there is also a reel
of recorded greetings from people around the world wishing the VoIP
Community a Happy New
Hi,
Curious, do many of you check out software or projects when they have
a live CD or does that make any difference to you? Does anyone know if
the general public (not reading this kind of list) is attracted to a
Live CD more than an Install one?
thx,
/r
On Sun, Dec 20, 2009 at 5:10 PM, jon pounder j...@inline.net wrote:
Live usb sticks are another matter (assuming your bios actually reliably
boots them) at least you can save your changes and pickup where you left
off the next time.
Excellent point, thanks!
/r
On Sun, Dec 20, 2009 at 6:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
However the writable storage also allowed us in our live CD to save some
minimal configuration on the media. We have a CD version and a USB
version of our live system, which are basically the same. The system
On Sun, Dec 20, 2009 at 7:18 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
I am not sure if this is even on-topic for the biz list
If in doubt, why not skip it and move on? I am asking people who offer
asterisk-related products and voip-related products as Live CD, such
as Asterisk
http://vuc.me
Kamailio, Open SER and Asterisk walk into a bar...
The bartender is Alex Balashov, someone whose posts I have long
admired on this list. Alex has agreed to take us through the following
areas:
- Relationship of Kamailio to OpenSER project history.
- What is Kamailio/OpenSER?
-
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote:
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
http://www.voipusersconference.org/2009/sip-for-apple-iphone/
I have not done any Asterisk-specific
On Sun, Dec 13, 2009 at 11:24 AM, Randy R randulo2...@gmail.com wrote:
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote:
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
http
On Sun, Dec 13, 2009 at 8:45 PM, meetmecall i...@meetmecall.nl wrote:
Siax is working great for me and as far as I know/remember well, you
can get it from the app store for a reasonable price. It supports SIP
and IAX2 and works easy with Asterisk.
It looks like it requires a jailbroken iPhone,
Hi,
We had a last-minute cancellation from Vivox for today's conference.
It happens that someone suggested a guest idea, Howler Technologies
CTO Jay Fenton, who agreed to join the call from the road. Anything
you want to know about transcoding to and from g729 is out topic for
the first hour. My
Hello,
I am working with several SIP projects that use g722, or are trying to
do so, with pjsip library.
According to pjsip team's interpretation of g722, it works with 14bits
PCM for input/output, so pjsip basically 'converts' the audio sample
from 16 bits to 14 when encoding and vice-versa.
On Mon, Dec 7, 2009 at 3:23 PM, Kevin P. Fleming kpflem...@digium.com wrote:
As far as I am aware, for ITU-T compliance the codec only cares about 14
significant bits, but the reference source code needs those 14 bits in
the *top* 14 bits of each 16-bit word that it supplies/produces. The
VoIP Users Conference begins in about 30 minutes to discuss the use of
VoIP on social networks like Facebook. If you have any interest in
this (or maybe you customers do?) please join us
IRC anytime: #vuc on Freenode
SIP see http://vuc.me for all the URI and PSTN numbers
Skype:vuc.me or
externip=123.123.123.123
On Tue, Dec 1, 2009 at 4:32 PM, Joao Gomes Pereira
gomespere...@startel.pt wrote:
Hello
I'm trying to register an Asterisk working behind Nat.
Here is the trunk:
register=username:passw...@sip.startel.pt
[startel]
type=peer
host=sip.startel.pt
username=username
On Fri, Nov 27, 2009 at 1:54 PM, Marco Cordeiro
marco.corde...@globalstar.com.br wrote:
Do you guys suggest any 1800 DID Provider in the US ?
We like OnSip.com / Junction Networks stable and various service
levels from none of hosted pbx. You should post this to the -biz list.
/r
On Tue, Nov 24, 2009 at 2:36 PM, jefferson alexandre
jefferson.alexan...@gmail.com wrote:
On Tue, Nov 24, 2009 at 11:21 AM, Jerry Geis ge...@pagestation.com wrote:
Is there a way to keep asterisk in RAM
and tell linux not to swap it out (ever).
On a closely related note, has anyone built a
On Tue, Nov 24, 2009 at 3:42 PM, Richard Kenner ken...@gnat.com wrote:
On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
And? Noticed any significant performance advantage?
/r
On Fri, Nov 13, 2009 at 11:31 AM, Manu et...@manu-dpk.net wrote:
Can you help me please?
Thank you very much.
Voici un meilleur site pour poser des questions de tout genre en français :
http://asterisk-france.net/
/r
___
-- Bandwidth and Colocation
If you missed @voicegal last time or didn't go to Astricon, join us
today on the Voip Users Conference to meet Allison Smith, the voice of
Asterisk.
Or go listen to the FBI talk about security...
http://VoipUsersConference.org for details.
/r
___
--
Hello from http://VUC.me or voipusersconference.org
This week on the VoIP Users Conference we welcome the Village Telco
Project [http://www.villagetelco.org/ ] self-described as an
easy-to-use, scalable, standards-based, wireless, local,
do-it-yourself, telephone company toolkit.
This is an
Hi,
If by chance you should find your self in Paris or wish to be there to
present... this is for you.
Note they do NOT want commerical presentations and this is only about
Open Source Asterisk
http://www.astrieurop.com/en/
I am considering going. Digium being a premier sponsor, I imagine some
Alex,
You forgot to clip the extra from the quote, shame on you!
On Mon, Nov 2, 2009 at 9:47 AM, Alex Balashov abalas...@evaristesys.com wrote:
Tzafrir Cohen wrote:
Top-posting, on top of your other sins.
Please spare us this capital punishment.
An entirely fair point.
Nevertheless, I
On Wed, Oct 28, 2009 at 5:05 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
Let's be realistic here. You need to 'drink the koolaid' before you
install it for a client. What I'm saying is you really need to install
Darrick,
No, he already drank the koolaid by believing in asterisk. Now
Tim Panton went to the pains of recreating his Astricon preso today in
the form of a screencast:
http://blip.tv/file/2762980
Amazing future of the Google Wave / Voice combo IMO.
/r
___
-- Bandwidth and Colocation Provided by
As others have said, John, Viddler is good. If you have any
shorter-than 10 minute videos, you might put them on YouTube as well
for the sheer exposure and then add something pointing to a Viddler
URL for additional, longer content.
/r
___
-- Bandwidth
On Wed, Oct 21, 2009 at 9:57 PM, SIP s...@arcdiv.com wrote:
Sounds like it wasn't a very interesting track. ;)
Not sure, but I guarantee the previous night was interesting :) The
VUC guys, sometimes led by Randal Happy Hour Schwartz, know how to
party. One night I got two hours sleep and was
On Wed, Oct 21, 2009 at 8:28 PM, Danny Nicholas da...@debsinc.com wrote:
Is THAT a summary :)?
As I said above (or below?) I we'll be talking about this on VUC
Friday at 12 Noon. In fact, here's the whole spa^H^H^H preview:
VoIP Users Conference (VUC) Astricon, Been there, Got the T-shirts
On Wed, Oct 21, 2009 at 4:01 PM, Bob Pierce pier...@westmancom.com wrote:
Or charge for full access! Leave a few teasers, and charge some amount to
see them all. I would pay - even close to attendance price... could only
help you get past break even ;)
I agree, I would be quite willing to
On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote:
Randy R wrote:
I missed the first part of this, but has anyone said: not all the
presentations were recorded.
Hi Randy.
Yes, that was mentioned. Actually, three of the four tracks were
videotaped IIRC.
Barry
On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
| I have three Snom M3s at the moment but getting pretty fed up with
| the
| issues :( I am UK based and would be interested to hear of other
| peoples
The S685IP has no headset jack AFAIK. If you want to use a headset
Looking at my shiny new Google Wave account, I was wondering if anyone else
on this list is in the beta AND going to Astricon. Astricon seems like it
would be a good test of the kind of collaboration GW is trying for. In any
case, I'd love to try to do an Astricon wave so let me know if you're
=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE
;CN=Randy R;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
CLASS:PRIVATE
CREATED:20080306T082859Z
DESCRIPTION:Every week we try to get guests with ideas\, products and servi
ces you
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