Questions ...
OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes,
and extensions. All exciting.
Two questions:
I'm in a natted environment and need to utilize a SIP provider to make
calls
in the US. Currently I have Vonage in my natted network and it works
fine,
UnixODBC. No need to rewrite everything for a simple DB change.
In what language is it written in? It would be interesting to at least
look at it and maybe convert it to use MySQL instead.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Ok.. Let me pose a question regarding this configuration.
Lets say you have the ISP bring in a full T1 and they split it half
voice half data. They would usually do this in a channel bank on
site... So in this scenrio... You have the Channel Bank from the ISP
where they split the channels.
Exactly.
So...
I would need as you noted two T100P cards or a T400P. The T1 goes into
the * Server and the second port of a T400P goes back to the asterisk
server. Then the extensions get broken out from the Channel bank?
Geoff
___
Either way will work. Getting the T400 four port card gives you room to
grow, but getting 2 T100P single port cards saves you about $500.
Is this the only way to handle extensions... This turns a 4 port T1 card
into a 2 port card... Is this the suggested method?
Geoff
Below you will find, what I believe to be a typical setup with a T100P
card. My question is -
1. Is this correct?
Possibly. Depends on if you use a channel bank that can do add/drop and
you're not using a PRI.
You'll take your incoming T1 and go into 1 T100P and use another T100P to
feed
Which one would one should I use to solve my problem? Does an loadable
application give you more control than an AGI script?
If you want something that runs continuously (such as a listener process
or control process), it'll have to be a loadable module. AGI scripts only
get run when the
It is that type of mechanism that enum uses and yes it was to solve a
similar goal, but in this case you need a 'route server' type system - in
particular as this is for IP routing of PSTN end points not on an IP
network.
A discussion about this came up a while ago. I suggested something
Actually, if this was to be done, it might be an idea to do it with DNS, so
client machines would just do
Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS
system would resolve which machine is the correct target - no cleverness at
all required at the client end, so implementation would
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've searched the mailing list quite extensively, but didn't come up
with anything promising (some things wer helpful, though). Does anyone
know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310)
phones can be made to work with
On Mon, 29 Sep 2003, Bill Leckey wrote:
I've been playing with the outgoing call spooling feature a bit lately
and it all works as it should with the exception of one irritation.
I'm mostly using SIP to talk to the phones and using G.723.1
I copy the call file into the spool/outgoing
Interesting that it has 2 ports on it, and a speaker. The picture looks
a whole lot like a modem to me.
The real X100Ps look like a modem too. They have 2 ports and a speaker.
When I misplaced mine, I rummaged around looking for it and kept finding
it but putting it back in the pile thinking
Hi,
Can anyone suggest a good motherboard for the T/E410P cards ? Coz it
doesn't get inserted in the the regular P4 motherboards due to PCI slot
(32 bit) Any suggestions.
I'm an AMD Athlon bigot, I'm using the MSI-6501 dual AMD MB. Its got 2
64-bit PCI slots that'll take a TE410P.
Do they fall under FCC certification if they're built to the same
specifications as the ones from Digium? If I build my own T100Ps from the
schematics and board layouts that are available, are they legal to plug
into the PSTN?
___
Asterisk-Users mailing
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including
On Wed, 2003-09-10 at 11:55, James Sharp wrote:
If I have a system with 1 machine to handle incoming H.323 calls and
then
multiple machines to distribute them to T1 ports over TDMoE, where does
the codec translation take place? Does it take place in the master
system
or does it take place
Is the max recommended still 2 cards, even in a Quad Xeon with
superduperwhizbang Hyperthreading? I'll be running incoming G.729 audio
out to TDM.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
So 3 or more TE410Ps in a system?
Is the bus mastering design that much of a significant improvement?
I would strongly consider the TE410P in this configuration and would be
interested in working with you to check scalability.
Mark
On Wed, 10 Sep 2003, James Sharp wrote:
Is the max
Hi all,
I've got myself all confused about the capabilities of *. I somehow
convinced myself (because I see a lot emails flying around about IP
phones)
that Asterisk works as a PBX and trunking gateway, but does not do voice
coding (i.e. TDM in, VoIP out). Does Asterisk work as a VoIP
allow this to happen. Do you know of any tools that convert ASF to
mp3?
mplayer/mencoder understands ASF, mp3 and lots of other formats.
Wont play an ASF stream, though...which is what he's looking for.
___
Asterisk-Users mailing list
[EMAIL
Wont play an ASF stream, though...which is what he's looking for.
you're sure?
e.g.
mplayer http://live.atlas.cz/radio1/radio1-32.asx
works fine here.
Well, hell. Make a liar out of me. It wouldn't last time I looked.
___
Asterisk-Users
[thread change, different topic]
is
How about a little tiny program that connects to a remote host, grabs
the contents of an MP3 stream, and pushes it into a FIFO locally? It
would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest
it as if it was a local file. The program
On Thu, 2003-09-04 at 17:22, Peter Pauly wrote:
Does the Digium FXS card support modems (and Tivo devices)?
If so, to what speed have they been tested?
Assuming that you can do native zaptel bridging (Going from an FXS port to
an FXO port in the same machine), you should be able to get up to
Again, not near my asterisk box so I can't check this out,
but is it possible to have the different ports drop into *
in a different context for each line? That way you could
just set up an 's' extension in that context for the
different attendants.
Yup. Set up different contexts in
On my SBC phone, I used to hear a high-pitched chirp before the Call
Waiting beep (much like the first chrip of a V.90 modem negotiation tone)
when someone called in and I was on the line. Does this mean SBC was using
FSK to transmit caller ID on my line?
Yup. That's CallerID over Call
Oh, and let's not forget that the traditional carriers are
not ignorant
of what is happening with VoIP or customer interest. There
is no doubt
that they are aware that if they don't find a way to deliver
this service,
someone else will.
No, if they don't find a way to deliver
Mike,
I opted for an integrated T-1 for 1 customer who needed about 12 lines.
I configured it with 12 lines voices and 768k data. Chances are you need
this kind of bandwidth if you need 12 phone lines. Combining it on 1 T-1
can make it a little more cost effective and of course one of
Its another one of my If I only had time...damn this sleep thing ideas,
but I really wonder how hard/cost effective it would be to build an open
source IP phone or phone adapter (ala ATA).
In about 20 minutes of mulling and research, I figure you could do it for
about $40 in parts plus coding
On Tue, 19 Aug 2003, Michael Sandee wrote:
I guess you will need some software/mem/cpu/flash too? getting it on a
cicuitboard etc?
Software would be opensource...get a couple of people together to write it
RAM I missed, I thought the C400 had onboard ram, but it doesn't...so add
another $10.
Could you tell me where mysql/errmsg.h is located on your
distribution? We can update the Makefile to look there for that
header.
Can't you use mysql-config to get the include and library paths? Granted,
you still need to make sure that mysql-config is in your $PATH, but it
keeps you from
There is the other hurdle of clients with existing PBX systems in place.
I've no idea how we'll cover this scenario as I'm sure most clients will
be
reluctant to replace their existing systems, unless of course asterisk can
be plugged into some of these systems?!?
Yes, it can. If the PBX
RE: [Asterisk-Users] newbie question - devicesHi,
So let me understand this better.
Asterisk can use SIP gateways which offer PSTN access. For example
www.iconnecthere.com, can be used?
Is this correct? And if it is, than any incoming calls through that
service, could be redirected by
Is anyone else having trouble accessing it with something besides IE on a
Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE
on Mac and Netscape on Solaris Linux explode when loading
login_page.php.
___
Asterisk-Users mailing
On Fri, 1 Aug 2003 15:25:49 -0500
McAughan, Matt [EMAIL PROTECTED] wrote:
Have you setup the zaptel.conf and zapata.conf configuration files for
how
ever many ports you have on the card and then run the ztcfg -vvvc
command?
Since the module aren't loaded, config zaptel.conf,
For the development team to get * (and the zaptel cards) running on BSD
shouldn't take too much effort. Perhaps it's just a matter of finding the
right incentive? My only request would be that it be installed to match
BSD
filesytem standards (everything in /usr/local).
One of my next
[EMAIL PROTECTED]:~# modprobe wcfxo
/lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device
/lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by
incorrect module parameters, including invalid IO or IRQ parameters.
You may find more information in syslog or the
Also, it isn't very easy to 'test' either, as the staff at the 911 call
centre won't appreciate your testing, and at least in Australia, it is
some
sort of criminal?/illegal offence to call emergency for non-emergency
situations.
I had much the same thoughts. Currently my 911 code is just
The way I've seen it done is that the incoming fax signal is digitized and
compressed, then sent over the IP channel. It is done in real time. You
end up taking up 7k-14kbps instead of the 32/64kbps you'd use to pass high
enough audio quality to not irritate the modems.
Unfortunately, this
Make sure you're using fxs_ks signalling for the FXO channels and also
make sure that your incoming lines support disconnect supervision.
Otherwise, * has no idea when the calling party hung up.
Hi Steven,
I have analog lines connected to the fxo lines of the Zhone channel
bank. All of your
parkext = #700 ; What ext. to dial to park
Try removing the #
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
101 - 140 of 140 matches
Mail list logo
