That's always been the site at that url.
On 8/2/05, Tony Hoyle [EMAIL PROTECTED] wrote:
Carlos Chavez wrote:
I have been using Sixtel from the beginning of the year and service was
getting worse and worse. Yesterday I tried to access the website to get the
CDR and I got an error
Greetings,
I have a Sipura SPA-1001. When I make outgoing calls, I have very
jittery sound. Incoming calls work fine. This wasn't the case a few
months ago, I am running head as of yesterday.
Any suggestions?
Thanks,
Erik
___
Asterisk-Users mailing
The Good:
I've had sixTel since December 2004. I've had surprisingly good call
quality and very short ping times. Overall the service has been good.
Response times for support were great when I needed it in the past. I
got a $10 rebate for my toll free DID because it was a couple of weeks
late,
All I'm saying is avoid sixTel. They have great sound quality, good
prices, good ping times, and a pretty control panel. However if you
ever need support anything done that doesn't happen automatically on
their web site (number doesn't ring, custom toll free did, refund,
support in general) forget
leak issue will be
resolved ?
Thanks
Walter
--- Original Message below ---
Message: 20
Date: Sat, 16 Jul 2005 21:42:44 -0700
From: Erik Espinoza [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Memory leak in asterisk CVS
To: Asterisk Users Mailing List - Non-Commercial Discussion
I once tried to call in support for digium for 4 IAXy's that I
purchased ($400). They told me to e-mail this mailing list. I
appreciate all the hard work that they did to produce asterisk, I just
don't trust this company to support anything. Just my $.02.
Erik
On 7/18/05, Andrew Kohlsmith [EMAIL
Known issue. This was reverted later.
Check the thread on the mailing list
http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html
Thanks,
Erik
On 7/16/05, Walter Klomp [EMAIL PROTECTED] wrote:
Hi,
My Asterisk CVS is apparently not doing much (other than keeping SIP
IAX2
Hello Rob,
I don't really know the answer to your woes, but I have a queston.
Where are you downloading the rxfax and spandsp? I can't seem to hit
any of the sites that claim to have it.
Erik
On 7/13/05, Rob Danz [EMAIL PROTECTED] wrote:
Hello,
Let me start by saying I have checked
I can only think of 2 ways to proceed:
1) Set a shorter register interval
2) Set static ip on all phones, and forgo registration
On 6/30/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I am using Linux-High Availability between two Asterisk servers, everything
is fine
Agreed. IAX2 would have been a much better way to go. Regardless i
don't see how an open source, standards based softphone will compete
with Skype. Skype has a few things going for it:
1) Hype, lots of it. It's no coincidence that that the two rhyme
2) Built in traversal of firewalls - p2p style
On 6/30/05, Michiel van Baak [EMAIL PROTECTED] wrote:
On 15:30, Thu 30 Jun 05, Erik Espinoza wrote:
Agreed. IAX2 would have been a much better way to go. Regardless i
don't see how an open source, standards based softphone will compete
with Skype. Skype has a few things going for it:
1
Nope, sixTel has been acting up big time recently. I've put in
requests for support online. Calling has gone to a message that says
go to the web site.
This really blows, the prices were good but unless they shape up I'm
switching providers.
Thanks,
Erik
On 6/27/05, JD Austin [EMAIL PROTECTED]
Hey does anyone know how to configure the 4 port fxo to work with
Asterisk? I have the updated firmware. All ports register, however
incoming calls are never handled properly by the fxo. I even set
hotline.
Does anyone have any info, or perhaps a web site?
They probably don't want to deal with the snickering and laughter that
this code will ensue. Digium has a good rep. in the open source
community for Asterisk, they don't want to release this mockery on
anyone!
On Wed, 23 Mar 2005 15:38:06 -0600, Chris Wade [EMAIL PROTECTED] wrote:
I've
remove auth=md5 from your iax.conf and try again.
On Tue, 22 Mar 2005 21:38:15 +0100, Androtech [EMAIL PROTECTED] wrote:
Dear All,
I bought one IP PHONE from Integrated Networks which was showed to wiki too:
http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
I have problems
Never had a problem with mine. I set my DHCP server to hand out a
specific IP to the IAXy, too.
Works on some, but not all. It fails on Microsoft and Cisco DHCP
Servers. Just because it works for you doesn't mean it's implemented
correctly.
What's the advantage over unplugging the unit and
The ulaw codec is the heaviest codec there is. Have you tried lighter
codecs such as the gsm codec?
Erik
On Thu, 10 Feb 2005 22:01:01 +0100, Michiel van Baak
[EMAIL PROTECTED] wrote:
Hi,
My experience:
A handfull of concurrent calls, all works fine on 54mbit.
Dont try to go beyond that.
Create two extensions on the phone, the 7960 allows this. Then
associate the second one with the sales mailbox.
Erik
On Thu, 03 Feb 2005 13:56:33 -0500, Martin Roy [EMAIL PROTECTED] wrote:
I'm wondering if there's a way it will show on the phone when there's a
new message. Here's what I'm
Sounds like you need a SIP Proxy, which just relays calls to the
destination, rather than Asterisk which handles all aspects of a call.
I'd recommend Sip Express Router, http://www.iptel.org/
On Thu, 27 Jan 2005 12:58:56 +1100, Mike Sander
[EMAIL PROTECTED] wrote:
Hi,
We are in the business
I have one of these phones. I bought it off of eBay. Not sure where to
get them direct. You will need to load the proper image, in that I
believe it ships with SIP by default. Each protocol has its own image.
Erik
On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver
[EMAIL PROTECTED] wrote:
Ouch, sorry to hear that you bought 20 of those damn things.
I'd recommend you provision them up with a static. The IAXy doesn't
really use dhcp, it uses a mix breed of bootp and dhcp which works on
only certain implementations of dhcp with varying degrees of success.
Since it is acting more
No, it only does bootp
On Wed, 19 Jan 2005 15:24:23 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Can IAXy be setup for PPPoE ???
If so, how?
bye
Ronald
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
What exactly do you want an unregistered phone to do? Ring nothing and
then go to voicemail? What is it that you are trying to accomplish?
On Wed, 19 Jan 2005 00:25:08 +0100, Rob Scott [EMAIL PROTECTED] wrote:
Asterisk seems to regard an unregistered phone to be busy.
Is that correct? Is not
Whoops, I meant auth=plain for a packet8 dta.
Erik
On Fri, 14 Jan 2005 16:52:06 -0800, Erik Espinoza
[EMAIL PROTECTED] wrote:
Under your sip.conf change to this:
[8006]
type=friend
host=dynamic
auth=md5
secret=1234
dtmfmode=rfc2833
context=sip
callerid=8006
[EMAIL PROTECTED
Did you enable passthrough for the rtp ports on the asterisk box?
I had the same problem until I enabled udp 1:2 on the firewall.
On Wed, 12 Jan 2005 22:06:18 +0100, Bruno Hertz [EMAIL PROTECTED] wrote:
Hi folks
an issue I don't understand. I'm running * stable 1.0.3 on public
The 2100 has two ethernet ports
On Thu, 6 Jan 2005 09:29:48 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I've found approximate same pricing for both. Sipura 2100 seems to have more
features...
What are differences between those two ?What about their reliability
(specially
Greetings,
I recently dumped my packet8 line in favor of sixtel + asterisk/sipura
spa-1000. It's working great, but I want to enable call waiting and 3
way calling. When i am on a call and get a second call, currently
things go straight to voicemail. It looks like sixtel is passing on
the extra
Most digital devices such as modems, fax machines and tivo's can not
be used without a lot of changes on VoIP.
I've seen success with TiVo when you use a special code to kick it
down to 14.4 kbps and use g711ulaw as the codec. I think your best bet
is to try to eBay the custom nic for the TiVo
Oh yeah, for reference try looking at tivocommunity.com on how to
configure your tivo for 14.4 or 19.2 kbps
On Thu, 6 Jan 2005 19:46:48 -0800, Erik Espinoza
[EMAIL PROTECTED] wrote:
Most digital devices such as modems, fax machines and tivo's can not
be used without a lot of changes on VoIP
I'm not entirely sure this phone supports sip. Have you tried building
the asterisk extra's and configuring it with skinny?
Erik
On Thu, 6 Jan 2005 00:37:32 -, Paul A Brown [EMAIL PROTECTED] wrote:
I am having all sorts of probs. It just won't connect. Anyone got any
example configs I
Or just get a couple of these:
http://www.ipeya.com/VOIP_Products.htm
(Specifically the 4 Ports FXO SIP VOIP-PSTN Gateway)
Available from eBay at a discount at:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=61839item=5741966868rd=1
And do it all without worrying about irq's or the
Check this out:
http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/
There's an article on how to use openvpn to encrypt data between two
Asterisk Boxes.
Should help, looks easy enough.
Erik
On Tue, 28 Dec 2004 16:57:11 -0800, Christopher Dobbs
[EMAIL PROTECTED] wrote:
This
I'd recommend Firefly by Virbiage. It's free and works on third party
networks with sip/iax2 support.
On Fri, 24 Dec 2004 00:27:28 +0100, Bruno Hertz [EMAIL PROTECTED] wrote:
On Thu, 2004-12-23 at 16:36 -0600, Michael Van Donselaar wrote:
iaxComm is Open Source, and currently runs on Win32
] wrote:
On Thu, 2004-12-23 at 15:50 -0800, Erik Espinoza wrote:
I'd recommend Firefly by Virbiage. It's free and works on third party
networks with sip/iax2 support.
Any specifics as to the why? I browsed their site and it made a good
impression, but I didn't try it yet. The phone itself
As much as I appreciate the work done by Digium on Asterisk, it
appears as though the IAXy is not ready for prime time.
1) IAXy has no security of any kind, anyone with iaxyprov can
reprovision your device without so much as a password!!!
2) The IAXy doesn't work with regular dhcp, it uses bootp
try auth=md5 under sip.conf
On Mon, 20 Dec 2004 18:35:53 -0800, Paul Austin
[EMAIL PROTECTED] wrote:
Hi,
Has anyone managed to create a setup with a Sipura SPA-1001 as a client?
Right now I can connect to the device by dialing the extension number
but when I try to connect from the phone
NuFone.Net has 800 toll free IAX termination. They have good quality
of calls once everything is setup and running.
I'd recommend starting with a low dollar amount into your account
first, until you have everything working. I went through about 20
e-mails back and forth via their request
Hey,
Does anyone know how to use the cisco router with an fxo wic with
Asterisk? I don't have enough space on this device to support an IOS
that supports sip or h323. Currently the only one signaling in there
says Cisco. I assume this is the skinny protocol.
Does anyone know how to configure
Did that, and still no. I'm wondering if this just wont work with a
windows dhcp server.
On Sat, 20 Nov 2004 03:29:42 +0100, Wilson Pickett
[EMAIL PROTECTED] wrote:
I can't seem to get this device to grab an ip from dhcp. We have a
Look at the list of DHCP on the server, and if possible
I can't seem to get this device to grab an ip from dhcp. We have a
working dhcp server (unfortunately it is on Windows), but I don't show
any leases requested by the iaxy.
Anyone have any ideas?
The ethernet and phone lines are plugged in before the device is powered.
Thanks,
Erik
I didn't see an IP Voice, but I did see an IP Plus and an IP/H323. I
thought the IP Plus would have it, but was concerned. I know that
Asterisk supports h323 so I was considering that one.
Anyways thanks for the feedback.
Erik
On Wed, 13 Oct 2004 21:53:25 -0700, Erik Espinoza
[EMAIL PROTECTED
Hello All,
I have a router with an fxo that I would like to tie into Asterisk. I
have the sip configuration down and used the example at
http://www.voip-info.org/wiki-Asterisk+cisco+FXO to create a config
for the Cisco router (3640). Although the router detects the vic, I am
unable to set the
Alright, I'd like to start out by saying that this is just a proof of
concept. The final configuration will include the purchase of a couple
of TDM400P's so all flames for using cheap Winmodems please divert
yourselves to /dev/null and don't waste my bandwidth or your time
responding.
I've
Sorry, I hit reply instead of reply all.
Erik
On Mon, 11 Oct 2004 16:03:51 -0500, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Mon, 2004-10-11 at 14:07 -0700, Erik Espinoza wrote:
Since every zapata card generates 1k interupts per second and needs to
be on it's own interupt, your
If reply all actually responds to the reply-to header and reply doesn't,
your MUA is broken.
There is no reply-to header being added in from the asterisk-users
mailing list, I double checked this by looking through other peoples
posts. The MUA works fine with mailing lists that actually add the
welcome.
On Mon, 11 Oct 2004 17:49:52 -0700 (PDT), Steve Edwards
[EMAIL PROTECTED] wrote:
On Mon, 11 Oct 2004, Erik Espinoza wrote:
Just want to test with real phone lines to ensure that they
work with our existing pbx before deploying with 8 lines using the
tdm400p's
Consider using a t100p
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