search - e.g. for 'randulo' :)
Once you found me, you should have been able to find the post where
I've put names of most of the VoIP USers COnference people. More then
added their own.
https://plus.google.com/104027218792812194992/posts/Xvnbp1YWf9K
You can have people in more than one circle
On Sun, Jul 10, 2011 at 11:07 AM, Steve Davies davies...@gmail.com wrote:
Thanks for that Gordon. What appears to be missing at the moment is
the ability to interface or collaborate with a group of 'strangers'.
You watch the stream to discover people but obviously that's a long
process. It will
On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Google Plus seems to be a walled garden.
Wait for the API.
:r
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On Sun, Jul 10, 2011 at 12:25 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Sun, Jul 10, 2011 at 12:17:52PM +0200, randulo wrote:
On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen
I don't see you on G+, are you there?
:r
On Sun, Jul 10, 2011 at 12:39 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
I don't see you on G+, are you there?
Me? You may see me there if it proves to be a federated service.
Tzafrir, I know you so I know you won't take this as a personal
insult. Why comment on something you aren't a
On Sun, Jul 10, 2011 at 7:48 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
I corrected a few factual errors on your part. Then I answered some
direct questions by you. But if you only look for feedback from the
believers, why do you bother asking here?
My bad, in that case. Apologies!
:r
Go ahead and lambast me for this post, it isn't specific to Asterisk, but:
G+ has only been open at all for a week and I already am chatting with
over 200 people who are into VoIP, Asterisk and all the rest of the
stuff we here care about. If you don't care or are anti-social, fine.
But you owe
On Wed, Jun 29, 2011 at 11:58 AM, Olivier oza_4...@yahoo.fr wrote:
Among TFTP, FTP, HTTP and others, which protocol would you select to
provision Polycom phones.
At the moment, I'm using TFTP but I'm wondering if I should pick something
else.
I use HTTP at home for better boot time and FTP is
On Thu, Jun 23, 2011 at 3:12 PM, Tim Panton t...@westhawk.co.uk wrote:
You should probably not mention the voipusersconfere...@gmail.com address
this for week's VUC
as at the moment the gateway ignores any calls to it.
If/when it comes back to life, we can realistically expect wideband
eComm is next week, Monday-Wednesday. Anyone here presenting? I know a
few names, Bryan Johns of Digium, for example.
If you are there, please stop by and talk to VUC as we will be there
live at the breakfast each day talking to participants. You can listen
to these chats or call in with
On Thu, Jun 23, 2011 at 1:58 PM, Tim Panton t...@westhawk.co.uk wrote:
The good news is that it supports a load of nice codecs now, including g722
:-)
And you know what that means?
:r
--
_
-- Bandwidth and Colocation
On Wed, Jun 22, 2011 at 10:23 AM, Administrator TOOTAI ad...@tootai.net wrote:
Le 22/06/2011 01:10, ERIC HERRON a écrit :
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text via a DID and it do something?
To do something with SMS to a DID, I'd recommend you take a
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
It's already qualified to win in the grammar and spelling categories.
/r
--
_
But Steve... didn't you just top post?
On Mon, Jun 20, 2011 at 10:52 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Two requests, not from me but the community.
1. Don't top post
2. When you find your solution, reply to this thread so others will be
(silver) spoon fed the answers and
Hi,
This Friday Chris Matthieu of SMSified.com will explain how to send
SMS from your apps. As usual, there will be talk about Asterisk,
questions, answers, and comments about telephony, networks, VoIP and
even some OT. All are welcome to join the weekly average of 35-60
callers live. If you
On Thu, Jun 16, 2011 at 11:52 AM, virendra bhati virbh...@gmail.com wrote:
If I am right then will you discuss about the sending sms with asterisk into
that conference ?
We can if someone wants to, that's how the VUC works.
:r
--
Hi All,
Today on the VUC, we'll be welcoming Sipgate CEO Thilo Salmon back to
tell about their choice of partners in their latest services.
I will be announcing the first #VUC VoIP Tell discount code for
Astricon in Denver, October 25-27.
Join us on sip:200...@login.zipdx.com using g722 or
On Fri, Jun 3, 2011 at 11:28 AM, devr devr d...@gmx.com wrote:
I am thinking about using numbers from voxbone. Before I make up my mind if
this is the right service for me I want to know what kinds of details will
be found when checking up on a voxbone number.
I am interested in UK numbers.
This Friday on VUC :
Part I : Grandstream Networks, they used to be entry-level phones, now
moving to video and surveillance cams
Part II : Junction Networks engineers discuss their move to IPv6 -
this should be compelling listening for all of us (Hi Olle!)
All welcome to join:
On Wed, May 25, 2011 at 10:53 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this
coming since the day Skype was first released?
Tim Panton, who's beenworking with SfA since it came out, posted this
article today:
On Tue, May 24, 2011 at 10:50 PM, Matt Darnell mattdarn...@gmail.com wrote:
We expect that users of Skype for Asterisk will be able to continue
using their Asterisk systems on the Skype network until at least July
26, 2013. Skype may extend this at their discretion.
It's widely believed.
Today, sessions 320-321 of the VoIP Users Conference will take place
at the usual time, 12 Noon Eastern [ http://vuc.me/next for local
times ]
We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You
can download or watch his accompanying slide presentation here:
On Tue, May 17, 2011 at 7:30 PM, Mike l...@net-wall.com wrote:
Is there any softphone or TAPI plug-in that allows one to dial from a web
page? As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application. I’d like to use this but on a
normal
This should be interesting, a double header Friday at 12 Noon EDT,
session 2 at 1PM EDT.
1) Pascal Doré, Media5corp. Pascal will talk about what they've been
up to in the year since his last visit. Thanks to the Asterisk mailing
list and VoIP community, their Media5fone was able to fix its g722
On Wed, May 11, 2011 at 6:47 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
I don't need a public or private Thank You When I was posting all the
time, I figured the ratio of Thank you emails to silence to be about 20 to
1, maybe as high as 50 to 1.
I agree with the others who are
On Wed, May 11, 2011 at 12:48 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Thanks Randulo,
I am surprised you noticed that.
I truly give thanks to all productive members of the Asterisk community.
Second that!
Would you say that I am a productive member of the list and go pretty
On Mon, May 9, 2011 at 9:47 AM, Jay R. Worthington
jayrworthing...@gmail.com wrote:
gateway for asterisk? I could not find any SIP-Gateway in the Market, and i
Portech has made GSM and CDMA gateways for years - nothing that works
with your old Android phones, though.
On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com wrote:
Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not?
:r
--
_
--
Hi all,
Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org
(formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz)
about Jabber, something the Asterisk community is becoming more
interested in by the day. Join us to learn more about Jabber and SIP
or to share your
Hi all,
You're welcome as always to join the talk on the VoIP Users
Conference, VUC for short. VUC began as the Asterisk Users
Conference but for obvious reasons, we changed the name in the first
year, although Digium was our sponsor for three years. We still have
plenty of you who are asterisk
On Fri, Apr 15, 2011 at 1:56 PM, virendra bhati virbh...@gmail.com wrote:
I want to join this conference but please tell me the topic of conference
and the process of joining step by step.
You're welcome to join us! All this information is at the top of the
main site: http://vuc.me
Please
On Fri, Apr 15, 2011 at 2:16 PM, Satish Patel satish...@hotmail.com wrote:
Is this online conf? Or are there archived files we can review?
There are over 300 recordings here:
http://vuc.me
:r
--
_
-- Bandwidth and Colocation
Hi,
Today at 12 Non EDT, Dan York will be with us to talk about the recent
on and off moments of Google Voice SIP URI calling. Like Skype +
Asterisk (or any SIP), Google Voice and SIP compose the other shoe
waiting to drop. We're following this with interest. So GV turned on
SIP URI and then a
Hi,
There's been a wave of questions on Asterisk lists about phones that
work well with Asterisk, services, etc. This week's VUC is all about
sharing your experience with various equipment and service providers.
The call begins on Friday at around 12 noon EST (9AM PST, 5PM GMT).
Info,
On Tue, Mar 8, 2011 at 1:51 AM, Dean Collins d...@cognation.net wrote:
http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html
Nice ;)
Hi Dean,
What I'm waiting for is when you can send GV calls to a SIP URI
without all the gymnastics
Hello,
This Friday March 4th is the 307th VoIP Users Conference. In a few
weeks, we'll be starting our 5th year of this weekly live event that
began life as the Asterisk Users Conference. We'd love to have you
join on on our call with this week's guest OnSIP.com by connecting via
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent A.
Torrenga
Sent: Thursday, March 03, 2011 11:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Provider Recommendation in US
I am becoming frustrated with our current VOIP provider. Does anyone have
On Tue, Feb 22, 2011 at 11:49 PM, Albert alber...@wp.pl wrote:
Yeah, this is messages which i saw before. Weird is that its hidden
somewhere under registration form and there was no notification about
cancellation for registered users.
Yes, it's in a popup when you try to register. I imagine
On Mon, Feb 21, 2011 at 11:56 PM, Albert alber...@wp.pl wrote:
does anyone know is AstriEurope coference is still on ?
http://www.astrieurop.com/fr/cloture.php
Cancelled.
Hello,
It is with regret that we announce you the cancellation of the
AstriEurop exhibition on May, 3rd and 4th 2011 in
On Fri, Feb 18, 2011 at 2:44 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
I'm VERY partial to Aastra's devices. Seriously, they don't take as long to
boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a
snip
Great info.
I do have a complaint about Aastra
Hi,
I'm excited to announce that the guys from AG Projects are stopping by
for a beer tomorrow on VoIP Users Conference, aka VUC.
You should already be familiar with their excellent multi-platform SIP
client, Blink (http://icanblink.com)
While Adrian and Saúl enjoy a few exotic brews with us,
On Wed, Jan 19, 2011 at 6:47 PM, Don Kelly d...@donkelly.biz wrote:
11:39 Parker said
That would fall under Quirk's Exception: Intentionally invoking Godwin's
Law to attempt to kill a thread is rarely successful. :)
Didn't work this time :)
Slightly OT: why is the Gmail ad server, which is
Also OT: Google combines message context with your personal search
history to do ad targeting, so look in the mirror.
I just made that up, though.
Not your mirror - your cookies!
No, it's true! Now I'm seeing Untimate Black Hat SEO (yes misspelled
because Ultimate was too expensive)
I was
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Greetings ${FellowVoIPuser},
When I saw the word Humbug in the Asterisk mailing list, I
remembered my friend Nir Simionovich had mentioned it to me at some
point, possibly at the big wine tasting party in Rostock during
AMOOCON, which may explain why I had forgot about it. Seeing the
thread on
2009/10/9 Juan E. Rodríguez jerdg...@gmail.com:
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
The Gigaset S675IP series of DECT/SIP phone has SMS capability, but
not sure it can work with Asteris.
/r
Quick reminder before Astricon (from which we will be reporting from live):
Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with
open source telephony solutions for the past four years, and has
collaborated on the development and growth of an international
Asterisk-based VoIP
This week Steve Sokol stops by to describe and field questions about
Digium's new affordable speech recognition solution. Later on in the
call, we'll also be looking at iVoIP, clients and uses for mobile
VoIP.
Join us on IRC anytime #voip-users-conference
During the conference, call via SIP g711
Hi,
Sorry about posted a protected link, I forgot we'd closed the site to
spammers since we don't use it anymore. The useful content was
re-posted in our list.
---
URI
Greetings,
We'll be getting together as usual at 12 Noon Eastern US Time for a
chat with David Duffet, a well-known member of the Asterisk community
and hopefully one or more of his co-authors of the new book Asterisk
1.4 Professionals Guide. In fact, I've been offered two ebook version
to give
Hi,
Take a look at this:
http://food4wine.ning.com/forum/topics/submit-an-application-for
Way down the page Dave VG submitted some scripts that hold the answers.
We also did a Polycom App conference at the VUC, but I can't find the
link right now.
/r
Hi,
This week we are pleased to welcome Andy Abramson
(http://andyabramson.blogs.com/voipwatch/) as our guest. Andy is one
of the most avid observers of the world of VoIP, from Asterisk and its
variations to all kinds of ramifications of VoIP. I'm sure we'll pass
a lot of the VoIP News in review
On Tue, Sep 15, 2009 at 1:28 PM, Jim Hankins
j...@allpointsmediaworks.com wrote:
Yes but I also have different hours for tue,thu and wed,fri. In the
o'reilly book
I have, it shows examples of using to group them, but I am getting
the error.
Not sure about the (what version?) but you can
On Tue, Sep 15, 2009 at 3:35 PM, Tilghman Lesher tles...@digium.com wrote:
That was my fault. My apologies, but this has been added in 1.6.2.
I suspected as much, since I couldn't find a single example searching
on the web.
/r
___
-- Bandwidth and
Most of you have needed at one time or another to sniff network
traffic for trouble shooting purposes.
Today I noticed that one of my SIP phone's web interface worked much
faster with Opera, so I wanted to see what exactly was going on. I set
up Wireshark and toook a look, but I got distracted by
Hi Gordon,
On Sat, Sep 12, 2009 at 6:04 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
It shouldn't reach out, but it will pass on ARP and other broadcasts
from Ethernet to Wi-Fi and back again.
I don't know what Free is - I'm guessing some sort of community mesh
network? Snooping
François,
Just to be clear, I am not on Free at all but with our mutual friends
at Acropolis Télécom. I see this Freebox traffic come out of nowhere
and it looks like someone trying to sniff my computer's wifi. A lot of
traffic is seen and I can't help thinking it isn't a good thing. I
should go
On Fri, Sep 11, 2009 at 4:18 AM, SIP s...@arcdiv.com wrote:
See... I would say the 'trapezoid' is one of the great strengths of SIP.
Which is why we need you all to come and discuss this, bringing up
other aspects, thoughts (even I have a few occasionally) and ideas.
@Dean - you will always
Hi,
We're pleased have a 25-year telephony veteran with us tomorrow,
Aswath Rao. Aswath maintains that Trapezoidal VoIP is Evil.
Join us and ask questions, make comments, argue about geeky details...
and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an
additional handset. Those of
Hello,
In the run up to Astricon [ http://astricon.net ] we'll be talking
today to Tim Panton about his experiences with SfA. You're welcome to
join in!
Speaking of Tim, you can join the conference in W I D E B A N D at
http://api.phonefromhere.com/gateway/zdx.xsql?conference=200901 - come
Tomorrow, Friday August 21st at noon EDT, we'll be on the rampage
again with the VoIP Users Conference, much of which concerns Asterisk
and the Asterisk community. We've covered a lot of ground in the past
2 1/2 years, and had a lot of great guest presenters.
In the meantime, there's a new gadget
On Thu, Aug 20, 2009 at 10:57 AM, David @ULCucoms2...@gmail.com wrote:
How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite
One way is to get a free test account at OnSip.com and create a SIP
URI. Then you configure IPKall number to call that SIP URI. Works
fine. You can't
On Thu, Aug 20, 2009 at 11:15 AM, SIPs...@arcdiv.com wrote:
IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one,
I'm not sure, but someone around has surely used it), etc, etc. There
are a lot of alternatives about.
Sorry, I forgot to mention IdeaSIP.com which works great,
Alternatively, get a SIP account with a proper ITSP and have then register
a number for you, then you just connect to them via SIP rather than have
them rely on connecting to you.
That would work well with the IdeaSip or OnSip solution among others.
There are many scripts to report a new IP to
I was about to post on this thread that I have contacted the makers of
iSip and they got back to me, we're working on a fix. We because I did
For info, the fix seems to solve the problem, VNET is waiting for
Apple approval on the new version of the app. I really like the
multiple accounts of
On Tue, Aug 11, 2009 at 1:57 PM, Philip A.
Prindevillephilipp_s...@redfish-solutions.com wrote:
Anyone have a chance to test any of the various iPhone SIP apps?
Here's a discussion of a few we've tried: http://VUC.me
I like iSip but the other two are good, too. iSip has multiple
accounts which
On Mon, Aug 10, 2009 at 4:53 AM, Alex Balashovabalas...@evaristesys.com wrote:
Word of advice: When you try SIP clients, focus on how the far-end is
hearing you, not whether you can hear them. In my experience, that's
where 90% of the deal-breakers lie with the iPhone.
Absolutely right! When
On Sat, Aug 8, 2009 at 12:42 AM, Enrique Moraem...@context.es wrote:
Finaly tried the WeePhone yesterday.
The WeePhone registered with Asterisk on the first try and call quality is
perfect with WiFi.
It's on the same LAN as the server.
For some odd reason, yesterday I tried weephone again
Enrique,
On Sat, Aug 8, 2009 at 12:42 AM, Enrique Moraem...@context.es wrote:
Finaly tried the WeePhone yesterday.
The WeePhone registered with Asterisk on the first try and call quality is
perfect with WiFi.
It's on the same LAN as the server.
Thanks for the extra info on the VUC.me site
On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashovabalas...@evaristesys.com wrote:
Which generation of the handset are you using? They differ in their
processing power and that may account for at least some of it.
Alex, this is just an iPod Touch, not even a handset. It doesn't have
a mic at all,
Ok, so now let me ask the question more directly:
I am looking for the best SIP application for the iPod Touch (Wifi
only). I don't care about 3g, Gsm or anything phone-related.
The app has to be able to register with an arbitrary SIP service
and/or dial arbitrary SIP URI. If it could dial one
So far, the best iPhone platform app I've found is a $10 one called
iPico. It is a one account SIP client, better designed than the others
and it actually works and can dial SIP URI.
I learned about it directly from Ruben Olsen mentioning it on the VUC
call an hour ago. I will be posting the
If you want to hang more results on this subject, please see the thread here:
http://www.voipusersconference.org/2009/08/sip-for-apple-iphone/
I'm very interested in anyone who is doing development in this space
so keep in touch. Basically, even though I've always preferred
DECT/SIP phones to
The subject of tomorrow's VoIP Users Conference will be mobile VoIP.
If you have any interest, please join us. I myself am tesing a bunch
of iPod applications to use with all the usual suspects: OnSIP,
Sipgate, Gizmo, Skype, your asterisk box, etc.
Details for joining the call are are at
Hi,
I've tried two SIP clients so far and both have unusable outgoing
audio quality. Skype app sounds fine, and recording the same mic
sounds fine, so I can only assume there is an issue with the clients
themselves.
Both clients allow you to register and make calls via SIP with any
abitrary
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Brunotipas...@gmail.com wrote:
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Could not generate Host-ID.
Make sure that you have eth0 enabled.
The MAC is used in the scheme to register and it looks like
Hi,
Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a
lot of experience in the telecommunications space and he joins us
today to chat about its current state, conferencing and whatever else
comes to mind. So we have a meta conference aout conferencing, it
won't be the first
Hi all,
You may have heard yesterday that Adhearsion and Voxeo have created a
new baby, Voxeo Labs. From our (non-biz) point of view, I'd recommend
following the blogs: http://blogs.voxeo.com/ to see how what they do
might be of interest to you and your asterisk/voip activities,
commercial or
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
It certainly would seem possible and would be a great feature request.
There probably is no circuitry existing to do it, but I would assume that
ohms, volts, or something could be measured while sending a
On Thu, Jul 23, 2009 at 5:08 PM, Danny Nicholasda...@debsinc.com wrote:
My .02 - IAX may not be an option and is probably not a good one if it is.
It requires a good bit of overhead to work reliably and well. You won't go
wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16
Please join us today at 9AM PDT, 12 Noon EDT for the VoIP Users
Conference to talk about the latest news and events in the wonderful
world of VoIP.
IRC #voip-users-conference
SIP 7463#2262...@proxy.ideasip.com for g711
SIP 200...@login.zipdx.com (for g722 wideband-capable devices)
See
Hi,
This week Tony Stankus, North American product manager of the Gigaset
line is our guest on VoIP Users Conference. I have had a two handset
S675IP in our small business for about a year now and my wife and I
both like the phone. But as a geek, I like it a lot more than she does
:)
6 SIP lines
(thank you gmail) so if you have DID without voice mail service, your
local Gigaset will handle the SIP channel as if it were
a PSTN line. This feature is selectable on a per account basis.
The phones also do g722 so they work with our ZipDX wideband bridge.
If you are considering new DECT
On Sat, Jun 27, 2009 at 11:06 AM, Olivieroza-4...@myamail.com wrote:
Hi,
Has anyone tried it ?
Is there any available pricelist ?
It is possible no one wants to answer this due to the NDA they had to sign?
___
-- Bandwidth and Colocation Provided
Though they have written me back twice to say coming soon I am still
waiting for the software...
So you'd rather have it even when it hasn't been finished?
Umm, no, but then when a company says looking for beta testers - please
sign up now! and then four months later has nothing to let me
Hi,
I met Matt Florell at AMOOCON and tried to record an interview. I was
pleased with the results, but later found that the battery deleted the
audio file when it went dead. Today, we'll have Matt live to talk
about VICIDIAL and answer any questions you may have about it.
For more on this:
Nir Simionovich is about to become a father. He will be joining our
conference at 12 Noon EDT today from the Maternity Ward to talk about
Amazon EC2 cloud computing with Asterisk. Nir gave a very good
presentation on this at AMOOCON a few weeks ago (see
http://www.amoocon.de for more on that). The
What happens if the http server is down? My point is that I don't
want it
to try and pull any config from a server. I just want it to use
its local
config.
I don't recall this looping probelm. The value of tries is supposed to
prevent this from happening.
r
On Fri, Jun 12, 2009 at 8:51 AM, Kengie Hokengiepa...@gmail.com wrote:
I am having some problems with Asterisk on static IP and Sipura-1001 on
dynamic IP. Is there any solutions to in the Asterisk configuration or
Sipura-1001 to re-register when the router change IP dynamic IP? Thanks.
If
Hi all,
In about 4 hours from this writing, the G.722 conference bridge will
be brought up, the Talkshoe G.711 also, so you can call in via SIP,
PSTN or Skype (experimental)
http://vuc.me for all the gritty details
IRC #voip-users-conference anytime today
We'll also be talking about hosted PBX
Already Friday, the week went by in a Flash.
If you haven't yet registered for a free Sipgate DID, I suggest you go
do so. If you are in, or interested in the business, Sipgate has a few
tricks up their sleeves and you should be aware of them. Someone from
Sipgate will be joining the conference,
How did I miss thins gem?
Polycom /will/ reboot on the drop of a hat /and/ take a damned
long time
to do it (~45-60 seconds) In addition, the web interface should
be
taken away and shot - the only real way to configure them is
through (T)FTP.
I didn't say
Hi,
Like me, some of you probably remember Jim as one of the pioneers
along with Leif and Jarod. These guys wrote the book, literally. Jim
is our guest tomorrow and he'll be talking about system building,
among other things. We always have a good time AND get stuff done on
the Conference so come
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not 3km.
That sounds really far for current loop stuff.
I was doing that too. I asked this same question a few years ago and
the answer was 100-200
On Thu, May 21, 2009 at 10:04 AM, Matt Darnell mattdarn...@gmail.com wrote:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
Not possible AFAIK.
2. Put Record on the main screen when a call is active. This would
eliminate having to press
Hi,
I met Michael Iedema of the Askozia pbx project at ANOOMA (ANOOMA is
the Cannes Festival of the Asterisk world) and begged him to take time
off from porting Askozia from FreeBSD to linux to tell us about his
project and the problems of porting as well. Hope you'll all be able
to stop by and
I caught Mark Spencer, Kevin Fleming, John Todd, Russell Bryant, the
other Mark in a truly Digium moment in Rostock, Germany on their way
to listen to the sea shanties.
http://tr.im/rawhide - be afraid, be very afraid
(Adhearsions' Jason Goecke is also in the picture somewhere)
/r
Zoa,
It was a pleasure to meet you! Please do come by some day, many people
would like to talk about your work and your client! Does it do SIP
URI? Call
sip:7463#2262...@proxy.ideasip.com
Best,
Randy
___
-- Bandwidth and Colocation Provided by
Anyone who was at AMOOCON and who would deign to join us (ahem, Zoa,
alors?) to hash out what happened and make fun of the presenters,
please join us Friday at 6PM Paris time (5 PM UK) or 12 Noon EDT.
I myself was really pleased to be there and meet so many interesting
and amusing people.
Some
Those reading the thread amy be interested in Askozia pbx
http://www.askozia.com/pbx/
Michael was at AMOOCON (great success by the way, thanks to all who
participated) and I was impressed. He will be a guest on VUC very
soon, possibly even this Friday.
/r
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