There was a trick for doing it posted to the Voxilla
(http://www.voxilla.com) forums. Here's the article:
http://voxilla.com/forum-viewtopic-t-1335.html
- |Daryll
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
The SPA-841 is either a 2 call appearance or a 4 call appearance phone
depending on whether or not you spend the extra $30.
In order to support call waiting, 3-way calling, etc. You must assign
both lines to the same extension. Fill in extension 1 so that it
registers with Asterisk using the Ext1
Outgoing caller-id seems to work fine. BroadVoice appears to send the
name that is on the account and the phone number. My dial plan uses
SetCallerID and SetCIDName, but the later is definitely ignored and
the former may not actually be required.
- |Daryll
Look for canreinvite=yes to get Asterisk out of the RTP path. Since
SIP traffic is infrequent and low volume having Asterisk in that loop
shouldn't be a problem, it's the RTP traffic you really want going
point to point. Realize that Asterisk can't get out of the loop if you
use the t or T option
Sounds like you've got a problem with your microphone. I got my
SPA-841 a while ago and the microphone works just fine. I don't have
to scream into it.
I like the phone a lot. I agree with you that the buttons have a
somewhat odd feel. They're sort of rubbery and don't slide like
plastic ones,
You probably want to use IAX to talk to FWD. It tunnels through NAT
without any special changes. See
http://www.fwd.pulver.com/advanced/iax
Making asterisk work through NAT is a pain and some of the Wiki stuff
is wrong/out dated. This works for me:
In sip.conf:
localnet:
On Mon, 7 Feb 2005 21:24:53 -0800, Paul Crick
[EMAIL PROTECTED] wrote:
Maybe something will change in a future software release..
The terminology between VOIP and POTS is different and unclear. If I
go in to an office supply store and buy a 2 line phone, it plugs in to
two wall jacks and I can
On Sat, 5 Mar 2005 15:02:47 -0700, Gabriel Gunderson [EMAIL PROTECTED] wrote:
May I suggest:
1) Updating your website that tells how to configure Asterisk for Broadvoice.
2) Answering emails to [EMAIL PROTECTED]
3) Emailing your users that signed up as BYOB when you think a change
On Fri, 19 Nov 2004 12:04:11 -0600, David Gomillion
[EMAIL PROTECTED] wrote:
According to the admin manual, the phone supports shared call
appearances (SCA) using the SUBSCRIBE-NOTIFY method in the 'SIP Specific
Event Notification' framework (RFC 3265). The events used are: -
'call-info' for
On Wed, 1 Dec 2004 08:31:15 -0500, David Cook [EMAIL PROTECTED] wrote:
I'm looking to give the SPA-3000 a whirl as I'm having too much
difficulty with the irq sharing thing inside the box.
I'm reading the book but without having one in-hand to play with it
appears a little obtuse at this
They don't seem to exist yet.
http://store.voxilla.com also lists the Sipura-841.
Atacomm is saying it'll ship in January. Voxilla says 2nd week of
December, but when I looked before Atacomm said the same thing, so it
may be that Atacomm has more recent information or just updated their
website
My asterisk box is behind a NAT firewall. I have friends that are on
Earthlink, Vonage, etc.
I'd like to make VOIP calls directly to them rather than going through the PSTN.
With Earthlink, I can make this work through FWD peeting numbers, but
that's sort of a waste of FWD bandwidth.
WIth
I've got soft phone that allows me to dial SIP URI's. I'd like to
route these calls through a provider to be completed, because I'm
beind a NAT box and doing it directly doesn't work.
Right now I've got an extension defined like this:
Dial(IAX2/${FWDUSERID}:[EMAIL PROTECTED]/**356username)
This
For what you want to do asterisk isn't really the right solution.
Asterisk is a PBX. It doesn't provide a way for you to connect to the
PSTN.
To do what you want, you need to buy VOIP service from any of the
providers (broadvoice, vonage, packet8, etc). They will provide you a
device that you
Because, the phone companies want to make money. They don't provide
all this infrastructure for free. It costs them money to run it and
maintain it.
No there is no protocol or way to access the PSTN for free.
Vonage is a phone company. They pay other phone companies every time
they want to take
I would guess IPKall is actually a CLEC. So every they get paid
settlements for every call termintated on their network. Therefore
having free incoming only lines is making them money.
The big difference is that they only offer Washington numbers and you
can't make calls out on their service.
If you mean phone service rather than a phone line, then your
statement isn't correct.
SpeakEasy has a service they call OneLink which allows you to get DSL
without phone service. It's an additional $6/month over their normal
DSL rates.
- |Daryll
On Tue, 12 Oct 2004 10:38:46 -0500, Matthew
I've noticed a similar problem. I've got an inline caller-id box that
shows message waiting and an analog phone that shows the number of new
and old messages. They both continue to show message waiting even
after I've listened to and deleted the messages.
So it looks like Asterisk isn't clearing
My message waiting isn't ever clearing.
You're using a SIP phone. So it sounds like Asterisk is signalling
that correctly.
Mine are analog devices connected through a Sipura-3000. Sounds like
the problem is related to the Sipura not Asterisk. Thanks for the
feedback.
- |Daryll
Working fine for me.
I installed their patch like they asked.
I'm registering with proxy.dca.broadvoice.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g.
A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the line, dial the new extension,
and flash again. At this
Digium is taking a some more equal than others sort of approach to
Asterisk. They figure that since they developed the base code, they
deserve a privileged position in the food chain, where they can do
things with the code that others can't. That is absolutely their right,
but I've never liked
On Sat, 2005-06-11 at 13:10 -0700, trixter http://www.0xdecafbad.com
wrote:
Look at 'big evil corporations' like apple. They did in a year with
mach what the FSF/GNU wants to do with HURD and still cant (to quote
stallman 'its really hard' while explaining why after 10 years HURD
still doesnt
I'm testing Asterisk 1.2. I read the UPGRADE.txt and followed the
instructions. I had to change a couple caller-id instructions, but that
was about it.
Then I noticed that voice mail wasn't working. It turns out my previous
config was calling VoiceMail2 instead of VoiceMail. That was easy to
It's really hard to secure an IP network once someone puts a hostile
server on the network. If you run ethernet to the rooms someone is going
to unplug the phone, plug in their laptop, and see what havoc they can
wreak. Ping flooding, DHCP servers, network/port scans, arp poisoning,
spoofing
25 matches
Mail list logo
