bx_config]
'8005095641' => 1. AGI(ivr-main.pl)
[pbx_config]
> check 'show dialplan nonauthenticated'
>
> regards
> Martin
>
> On Fri, 21 Nov 2003, James Sharp wrote:
>
>> I've got a couple of PRIs coming in from a SUM
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800
numbers routed through it.
When the calls come in, I get the following message on the console and the
call never makes it through:
(800 number is fake)
Extension '8005551212' in context 'nonauthenticated' from '232102749585'
>
> Hi all,
> We are thinking of changing our Nortel Meridian PBX to Asterisk. Before
> we
> jump into this we would like to know if we can support some important for
> us
> functionalities on Asterisk. We would like to know if we can
>
> 1. Have menu based voice mail with Aster
> 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 wor
> On Mon, 13 Oct 2003 14:01:00 -0400, Andrew Kohlsmith wrote:
>
>>> I really wouldn't like to run a telecom system on Windoze in the first
>>> place..
Last place I worked, we had to reboot our phone system every friday
night...it used NT.
___
Asterisk-U
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 Bar
> 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 chann
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
>
_
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
___
Asteri
> 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
fee
> 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 wou
> On Wed, 2003-10-01 at 16:40, James Sharp wrote:
>> I've got a handful of T1s going into a TE410. When I place calls into
>> the
>> system over these T1s, the system either doesn't decode all of the DTMF
>> digits or it decodes ones that aren't the
I've got a handful of T1s going into a TE410. When I place calls into the
system over these T1s, the system either doesn't decode all of the DTMF
digits or it decodes ones that aren't there.
When the system places calls out, there is no problem doing the DTMF
detection. Everything works great.
> -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 w
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/outgoi
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?
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Asterisk-Users mailing
> 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 TE41
>>
> 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 think
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, includin
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
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.
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> If the remote ends can do the codec, then yes. If they can't deal with
> the incoming codec, then it will be done at your h323 end point. The
> benefit of IAX2 trunking is to cut down on your ethernet load and to
> make expanding easier. Not to mention IAX2 is much better tested than
> TDMoE.
C
> 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
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 in each of the slave TDMoE systems?
Also, any idea how
> 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 Vo
>> 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-User
>> 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.
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[EMAIL
>
> [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 progr
> 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
> 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 i
> > 1. Will Asterisk route from one T1 to another "perfectly"? That
> > is, the bits that arrive on the Portmaster would need to be the
> > exact bits sent on the PSTN T1. Seem obvious that this should be
> > so.
>
> As of this weekend it does.
Can you DACS with it or is it just a passthrough t
> 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 Wa
> 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
> > 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
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.
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 codin
>
> I always have a chuckle when I see this.
>
> it probably could if someone sorts it out, but its reqally starting to
> expect a lot.
It'll just take someone with the masochistic tendencies needed to do the
realtime DSP code for reception. For transmission, however, things are a
bit simpler. T
> 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
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 lis
> 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
> 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
> == Parsing '/etc/asterisk/zapata.conf': Found
> WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No
> category context for line 1 of zapata.conf
zapata.conf needs to start with the line
[channels]
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> 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
> 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
>
>>
>>
>>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
> > Bumping calls to clear a path for 911 is possible within Asterisk
> already - see the "SoftHangup" application.
> That sounds good, but what can trigger the SoftHangup app to drop other
> calls automatically when 911 is dialed?
A short AGI script, perhaps?
_
>
> Too bad noone makes a cheap ethernet FXO. Why is it always FXS... :-/
>
I've been wondering how hard it would be to make a cheap FXO device out of
a Dallas Semiconductor TINI board and a TI DSP56000 chip.
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[EMAIL PROTE
> On Fri, 2003-06-06 at 16:09, Dante Alzamora wrote:
>> What is the best cost effective solution for a small office:
>> I need 3 FXS & 2 FXO.
>>
>> Can I hookup a TDM400P and 2 X100P on the same computer?
>>
>> Also, I saw some IP phones for $25.99
>> http://www.wosmile.com/cgi-bin/view_store_item.
> On Sun, 1 Jun 2003, Gene Kochanowsky wrote:
>> I saw that Digium now has a 4 port FXS card. Any plans to add a 4 port
>> FXO card?
>
> I'd like to see a USB FXO...
>
With as much weird and wacky stuff that people have been seeing with the
USB FXS (like totally not working, working only after a r
> On Thu, 2003-04-03 at 10:55, Mark Spencer wrote:
>> > 1. it was FCC, CA and CE certified (FCC and CA states no card is reg
>> with them as of last week)
For the T100P and T400P:
http://www.part68.org/tte_details.cfm?cicHistid=36439
http://www.part68.org/tte_details.cfm?cicHistid=36442
They wer
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 takes
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 y
> parkext => #700 ; What ext. to dial to park
Try removing the #
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> The only problem I can think that you would have with the
> ztdummy would be that to used a kernel source other
> then the one your running when you build it...
Or its not playing well with your USB hardware, which is what ztdummy uses
to generate the 1Khz interrupts that zaptel needs.
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