Re: [Asterisk-Users] SysMaster and GPL Violation

[Tom Lahti]

[snip]
If someone believes that they are contributing software to a GPL'd
software project, and does not realize that the nature of your disclaimer
allows Digium to release their changes under a non-GPL'd license, then
that is breaking with the spirit of the GPL.
If that is true, then the GPL is not comprehensive enough to cover its own 
"spirit", so what you are saying is that the GPL implementation is 
fundamentally flawed.

If one can break the "spirit" of it without breaking it, then something is 
missing from it that should have there.

On the other hand, if you are injecting some supernatural "spirit" (and 
purposely using that word to conjure imagery of the imaginary intangible 
qualities that can never be written on paper) of your own into what the GPL 
actually is, then the GPL is fine as written, which I suspect is the 
case.  The GPL is what it
says, and its spirit comes from what it says, and there is no way that 
anyone can break its spirit as such.

Unless you are now claiming to be the author of the GPL, you should stop 
trying to be an expert on its "spirit".  The only ones qualified to do so 
are John Stallman and his attorneys, misguided though they may be.

Yet no matter how much I don't care for the GPL, I find myself believing
contributors who don't fully understand the disclaimer merely to be naive,
but Digium looking a bit unscrupulous in this regard.
Butter him up and then call him unscrupulous in a later 
paragraph.  Beautifully manipulative.

That obviously won't fix the "moral standing" problem that the FSF would
Your own use of quotes here suggests something interesting.  I'll leave it 
to the reader to discover what.

--
Tom
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Re: [Asterisk-Users] BRI in the US

[Tom Lahti]

[snip]
I get PRI (23b+d) from Broadwing in Seattle area for $338/month, but we're 
colocated at their switch.  If you want loops out to your own location add 
$600/month or so to Qwest for the T1 loops.

--
Tom
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Re: [Asterisk-Users] CNG Comfort Noise Generation

[Steve Underwood]

Hi Assaf,
Assaf Benharoosh wrote:
I have a problem with many phone such as BudgeTone, ariaVoice, 
PCPhoneline. They are not generating comfort noise (you can hear 
yourself when you're talking)- with budgetone having CNG sporadically.
 
Is there a way to make this happen on Asterisk - or it must be a phone 
feature.
 
Does anyone else experiencing this issue with those phones and have a 
workaround?
 
Assaf Benharoosh
Hearing yourself when you talk is not comfort noise. It is sidetone. 
Comfort noise is simulating the background noise of the room at the far 
end when nobody is talking and transmission has stopped. Sidetone is 
always a phone feature. Comfort noise usually is too.

Steve
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[Asterisk-Users] random echo on TA750

[Paradise Dove]

all i have is random echo
I have already 4 TA750 with full FXO
echocancel=yes and echo training=800

- what should i do? 
- could it be solved with tweaking echo params on *?
- is there any additional devices that can be added between Channel
Bank and * to get rid off echo forever?

any help would appreciated

Paradise Dove
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Re: [Asterisk-Users] BRI in the US

[Joe Greco]

> Scott Stingel wrote:
> > Brian-
> > 
> > I was quoted (verbally) something on the order of $60 per month for a single
> > BRI by SBC in San Francisco about 60 days ago.  I thought that was high..
> 
> It has been a while, but the last BRI I ordered from BellSouth/Louisiana 
> was about US$109/month.

We're paying about half that per BRI (SBC/Ameritech/Wisconsin Bell).  I'll
note that at least up here, pricing is very dependent on whether you use
an ISDN ordering code ("package") - ordering a circuit with the same
features as a package can be twice as expensive as ordering the package.

This pretty much puts you outta luck if what you need isn't offered as a
standard ordering code.

It's mildly more economical (and, yes, as someone else said, so much cooler)
to bring in one BRI than two POTS lines with modest features like CID.

Are you guys actually able to make US BRI's work?  I'm interested in
hearing more...  we've been bringing in dialtone on ISDN for years because
we're in an RF-intense area, but right now bridging the gap from ISDN to
VoIP is a Netgear RT338 and some Sipura SPA3000's, and that setup doesn't
work half as well as I'd like.  I'd consider spending the money on a Cisco
with BRI VIC cards if I knew for sure it'd work well, but I'd be more happy
staying with a non-propietary solution.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> I hope this is the last time I have to comment on this...
> 
> > I'm struggling to think of another free software project where contributed
> > code bearing an identical GPL or BSD license would require any such
> > additional disclaimer.
> 
> How about GCC, GLIBC, etc?  Do you think Digium came up with those 
> disclaimers on our own?  No, we used the FSF ones with the exception that 
> the "long version" of ours has the original author retain copyright with 
> Digium as unlimited licensee, whereas the FSF requires copyright 
> assignment to the FSF and an unlimited license back to the original 
> author.  This distinction is because the FSF wants to be able to enforce 
> the GPL on code they did not author, whereas with Digium and Asterisk we 
> are more concerned that we do not allow contributions to pollute our 
> ability to make licensing decisions.

Hello, Mark,

Correct, and that is a fundamental ideological difference.  This works
for the FSF because their intent is merely to enforce the GPL.  You're
trying to do something else with it.  Those are not the same thing,
regardless of how similar the text implementing them is.

> Again, as with the FSF projects, nobody is *required* to make any such 
> disclaimer -- only if you want your changes to be included in what we 
> distribute as Asterisk.  By and large, I think people are willing to do so 
> because Digium has worked very hard to keep adding value to Asterisk and 
> because we've stuck to our guns on being sure that Digium created code 
> always remains Open Source and available under GPL.  Personally, I am very 
> much aware of the trust the Asterisk community has given to Digium, and 
> likewise I am very much aware the security and the independence *from* 
> Digium that GPL gives them.  Our future is totally dependent upon our 
> ability to continue to earn the support, help and loyalty of the community 
> at large.

I don't think any of that's at debate.

My *particular* concern is that people who would like to contribute to
Asterisk are told that they have to sign "this here disclaimer", and many
coders have not spent tens of thousands of dollars dealing with IP lawyers,
so the finer points (especially of the short disclaimer) might escape them.

It is counterintuitive that a project frequently referred to as a GPL'd
free software project would in fact have a sponsor that is licensing the
code under non-GPL terms, presumably for profit.  Messages on this very 
list in the past few months have suggested that some people prefer the 
GPL precisely because they believe that this should not happen in a GPL'd
project.

As someone who might have contributed changes to Asterisk, I did do a
little inspection to determine how hard it was to get patches committed
to Asterisk, because I had a few minor adjustments to suggest, but I was
a bit shocked to run into these disclaimers which are required for
contributions to the Asterisk tree.  I would have had no problems
releasing such contributions under a BSD-style license (which is GPL-
compatible), which would normally be sufficient to most other free
software projects.

However, the specific item that stopped me was the second paragraph of
the short disclaimer, because our lawyers would never allow signing of
a blanket statement such as "and will do nothing to undermine it in the 
future".  (As it was, the remainder of that paragraph would have had to
have been sent off to the lawyers, as I don't really have a grasp on how
much legal territory that might cover).

That sent me off to look at the long disclaimer, at which point it 
eventually became apparent what you were actually trying to accomplish.

Now, that's all well and fine, you obviously /can/ do it, but what most
disturbs me is that people might sign the short form agreement without
understanding exactly what it is that they're agreeing to.

If someone believes that they are contributing software to a GPL'd
software project, and does not realize that the nature of your disclaimer
allows Digium to release their changes under a non-GPL'd license, then
that is breaking with the spirit of the GPL.  The FSF writes two points
beginning at:

http://www.gnu.org/licenses/gpl-faq.html#ReleaseUnderGPLAndNF

which address issues relating to this, though I also have the opinion 
that Stallman has spent too many years in a cushy tenured academic 
position to be making such sour statements.

Yet no matter how much I don't care for the GPL, I find myself believing
contributors who don't fully understand the disclaimer merely to be naive,
but Digium looking a bit unscrupulous in this regard.

Is there any way that we can prevail upon you to at least fully disclose
what is going on to prospective contributors, perhaps in your disclaimer
files?

That obviously won't fix the "moral standing" problem that the FSF would
describe, but there's a certain type of moral standing to be gained from
open disclosure of your intent and disclosure of the actual purpose of

RE: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[Geoff Nordli]

[EMAIL PROTECTED] <> scribbled on :

>> So, if anyone is interested, I am suggesting particularly a
>> standalone, cross-platform project that is simple to install,
>> configure, operate and manage. It should operate with or without a
>> database. It can leverage existing projects, but it must not have
>> the existence or installation of those projects as prerequisite. In
>> other words, if this project uses another project's code, it must
>> also include the installation and configuration of that project in
>> this one's installer. 
> 
> I have an idea about that and I've been fooling around with
> some code to
> that end. My idea is similar to the Perl op_server.pl scripts except
> the managment interface is broken off into a proxy:
> 
> Asterisk <--(telnet)-->Proxy server<--(xml-rpc)-->Managment app
> 
> I have some crappy crap running in .NET that telnets to the Asterisk
> box and XML-RPC's the output to a client; even though .NET is Windows
> only (sort of - Mono) it's a safe bet that Asterisk is going to be
> deployed in a Windows-like environment 90% of the time, also you
> avoid GUI problems that you see with Java and GTK-Windows style apps.
> 
> I supposed astpy and the Twisted framework would address
> cross platform
> issues but you would have the GUI problem.
> 
> my 2c cdn.

If you want to go cross-platform then http://www.wxwidgets.org/ or more
likely something like http://www.wxpython.org/.

Of course you are going to have a gazillion different opinions on this.  

Geoff

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Re: [Asterisk-Users] Newbie question

[giovanni.powell]

i think u missing the defaultip of the phones.. e.g.

[general]
port = 5060 
bindaddr = 0.0.0.0  
context = from-sip
disallow=all
allow=ulaw  

register => 500460:[EMAIL PROTECTED]/500460

[fwd]
type=friend
secret=cuco99
username=500460
host=fwd.pulver.com

[2000]
type=friend
username=2000
secret=cuco100
host=dynamic
context=from-sip
defaultip=ip address of the phone
mailbox=100

[9001]
type=friend
username=2001
secret=anselmo
host=dynamic
context=from-sip
defaultip=ip address of the phone
mailbox=101




On Fri, 12 Nov 2004 17:43:38 -0600, Julio Tejera
<[EMAIL PROTECTED]> wrote:
> 
> Hello:
> 
> First, I'm really new to asterisk and I'm testing it in order
> to improve my first steps...
> 
> Recently I installed * asterisk on a FreeBSD Box (5.2.1)
> I got it working on my internal LAN (it works fine !).
> 
> I was trying to connect my * box through FWD using SIP
> but it is not working an I'm very confused about *, in fact
> I can't call from my * client (X-Lite) to a FWD number,
> but bettwen my * sip authenticated clients yes...
> 
> Please somebody can help me or guide me to the right direction ?
> 
> Any kind of help will be appreciated and excuseme by my english :o(
> 
> Here is attached my config files
> 
> 
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>
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> I apologise.

No problem.  It's easy to get all worked up over licensing issues, depends
on your politics.  :-)

>From my side, I'm kinda sitting here in amazement, watching how a list that
was so GPL-advocate a few weeks ago during the last license debate turned
so "who-cares-about-the-GPL,-let's-back-Digium's-position".

Especially since we had a very interesting segue in that discussion that
led into a debate about RHEL, and how companies might try to circumvent the
intent of the GPL while abiding by the letter of the license.

I don't have anything at all against Digium, of course, and certainly we
all appreciate Mark's efforts.

However, I do feel that the intent of the GPL is important.  It's an
interesting idea, probably even a great idea if it was an ideal world,
where no one had to code for a living.  In this case, I do see the 
intent of the GPL being bent, and I believe that there are questions 
that should at /least/ be asked.  To hide it all away from the light of
day is unfair to new contributors.  To pull it out in the open and
examine it is a reasonable thing to do.  We don't need to end up liking
it, after all, but knowing what is going on is still important.

So, for that, I apologize as well, but I simultaneously submit that I
believe it in everyone's best interest to be aware of the variance from
typical free software practices, and to think about what that means.

> Can we establish a "GPL Viloation" list somewhere so this can be  
> continuted there?

Well, if nothing else, we can all certainly agree that the original subject
matter is a most heinous violation of the GPL.  :-)  

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] Kirk IP 600 DECT station

[Julien Goodwin]

On Fri, Nov 12, 2004 at 11:06:52PM +0100, Remco Barende arranged a set of bits 
into the following:
> OK, I just got the Kirk IP 600 kit :)
> 
> It turns out that they actually make one unit that does H323 and can do 
> Skinny (Cisco) if you buy the version with a license for it. Mine supports 
> both.
Neat! Any info on pricing?

> The box appears to be running linux btw :)
> nmap revealed 2 open ports, telnet and http and also
> TCP/IP fingerprint:
> SInfo(V=3.55%P=i686-pc-linux-gnu%D=11/12%Time=419509CD%O=23%C=1)
> 
> I couldn't find any info on this station and how to connect it to * but 
> that's ok :)
> 
> It seems that you must configure the phones as Cisco 7940 in Call Manager. 
> The Wiki about the 7940 uses SIP which will not work so I have to try 
> skinny or H323.
> 
> I think the best way would be to use the skinny protocol but I'm a bit 
> lost there. When looking for info on cisco protocol I actually found 3 
> channels : chan_skinny / chan_sccp and asterisk-sccp.
There's:
* chan_skinny - basic support, supplied with asterisk
* chan_sccp - More phone support, more features, chan-sccp.sf.net
There are various splits and forks of chan_sccp out there, but the one
being activly developed is the one hosted on sourceforge. Asterisk-sccp
appears to just be the name that gentoo gave chan_sccp in their archive.

> Hopefully chan_skinny which comes with * is ok?
Should work for basic tasks, but chan_sccp supports more features.

> The wiki SCCP-HOWTO2 says :
> noload=chan_skinny.so
> but I guess I do want to load skinny?? should I specify load= ?
No, that's when using chan_sccp so that you don't have two modules
competing over the skinny port.

> I will keep posting results to the list and finally add a wiki when it's 
> working, hope any of the cisco experts can help :)
If you have any problems with chan_sccp drop me an e-mail.


pgpBVTD8I2bld.pgp
Description: PGP signature
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[Asterisk-Users] Cisco 7940 multiple line capability questions...

[Ian]

While I wait 2+ weeks for the shipment from cdw that'll allow me to log
into the Cisco web site to download the SIP firmware, I'm currently
trying to use a Cisco 7940 phone with the skinny protocol.


So far, I'm able to dial out from it just fine, the problem is
answering "outside" calls with it.  To be exact, the phone rings just
fine, I just can't actually pick up the call.


When I do a SIP to Skinny "internal" call within my own network, things
work as expected:


   -- Executing Dial("SIP/2001-b273", "Skinny/[EMAIL PROTECTED]|20|tr") in new
stack

Found device: 7940

   -- skinny_request([EMAIL PROTECTED])

   -- Skinny cw: 0, dnd: 0, so: 0, sno: 0

   -- skinny_call(Skinny/[EMAIL PROTECTED])

   -- Called [EMAIL PROTECTED]

   -- Skinny/[EMAIL PROTECTED] is ringing

Recieved Open Recieve Channel Ack

   -- Skinny/[EMAIL PROTECTED] answered SIP/2001-b273


The config responsible for this in extensions.conf is:

exten => 2002,1,Dial(Skinny/[EMAIL PROTECTED],20,tr)


Now, *sometimes* when I do a call from the outside in,
the call "rings" on 7940-1 instead of 7940-2.  This causes the phone to
pickup and hangup instantly when I attempt to actually answer the call.


I thought I could fix it by doing this:

exten => s,1,Dial(Skinny/[EMAIL PROTECTED],20,tr)


but I get this error from asterisk instead:

   -- Starting simple switch on 'Zap/1-1'

Nov 12 19:32:32 NOTICE[8731]: chan_zap.c:5436 ss_thread: Got event 2
(Ring/Answered)...

   -- Executing Dial("Zap/1-1", "Skinny/[EMAIL PROTECTED]|20|tr") in new
stack

Nov 12 19:32:32 NOTICE[8731]: chan_skinny.c:2515 skinny_request: No
available lines on: [EMAIL PROTECTED]

Nov 12 19:32:32 NOTICE[8731]: app_dial.c:777 dial_exec: Unable to
create channel of type 'Skinny'


So I'm wondering if there is a correct way to get asterisk to call one
line or the other on a single device?  Also, in the skinny.conf file,
is it possible to configure multiple lines for one device?  So far
everything I've tried just manages to write over the previous settings
for the first line.


Any help, or links to more information on this type of thing would be
appreciated.  I've read through the asterisk document project, and
googled for "skinny_request: No available lines" without results.



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Re: [Asterisk-Users] BRI in the US

[Mike Boger Jr]

Here I go lazily top posting:

I pay around 55 bucks a month for the circuit. I noticed you had what
appeared to be a quote for a PRI  in an earlier post : 23b+d is a heckuva
lot more expensive for sure than 2b+d.

Regards,

Mike
- Original Message - 
From: "Brian West" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Friday, November 12, 2004 9:41 PM
Subject: RE: [Asterisk-Users] BRI in the US


> One goal is to get BRI support in Zaptel if possible.  I'm right now in
the
> planning stage :P  Plus BRI is much cooler than pots.
>
> bkw
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Mike Boger Jr
> > Sent: Friday, November 12, 2004 9:05 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] BRI in the US
> >
> > Brian,
> >
> > Are you not able to get BRI where you are?
> >
> > Maybe the better question is what are you trying to acomplish? I've got
a
> > BRI in the house (with the horribly expensive Eicon sh*t that I paid 30
> > bucks for on ebay)
> >
> > Mike
> >
> > ___
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Re: [Asterisk-Users] The BV patch: Some notes

[Michael Wareman]

Hmm..  Where do I get this patch from?  I can find nothing on the Wiki
or on bugs.digium.com..

Of course - I may just be blind..

Thanks!


On Fri, 12 Nov 2004 10:43:20 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote:
> 
> 
> > Some notes on the Broadvoice patch
> >
> > * It makes your asterisk send less packets to Broadvoice by re-using
> >the authentication when re-registering
> > * It, by mistake, adds extra logging that is simple to remove from
> >your chan_sip.c. This was added to help debug the code, but should
> >not have been included in the distributed patch.
> > * It may also help other providers, but very few have such a limited
> >registration timeout. BroadVoice's equipment sets the expiration time
> >to a very low level to keep the NAT open.
> > * We will issue an updated patch during the weekend
> >
> > This code will be submitted to the bug tracker so it, if approved, can be
> > included in cvs head.
> 
> Olle,
> 
> Based on what you know and have experienced, would you care to guess at
> whether this patch should be applied to an * server that has a registered
> IP address?
> 
> I'm toying with the idea of waiting for the patch to be incorporated into
> the cvs and not trying to maintain special patch levels, etc.
> 
> Rich
> 
> 
> 
> 
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Re: [Asterisk-Users] BRI in the US

[Eric Wieling]

Scott Stingel wrote:
Brian-
I was quoted (verbally) something on the order of $60 per month for a single
BRI by SBC in San Francisco about 60 days ago.  I thought that was high..
It has been a while, but the last BRI I ordered from BellSouth/Louisiana 
was about US$109/month.
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RE: [Asterisk-Users] BRI in the US

[Brian West]

One goal is to get BRI support in Zaptel if possible.  I'm right now in the
planning stage :P  Plus BRI is much cooler than pots.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mike Boger Jr
> Sent: Friday, November 12, 2004 9:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BRI in the US
> 
> Brian,
> 
> Are you not able to get BRI where you are?
> 
> Maybe the better question is what are you trying to acomplish? I've got a
> BRI in the house (with the horribly expensive Eicon sh*t that I paid 30
> bucks for on ebay)
> 
> Mike
> 
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Jeremy McNamara]

Greg Boehnlein wrote:
Wow.. this is totally unacceptable. Has anyone contacted the GNU 
Foundation about this?

What has Sysmaster's response been?
Pure denial of the fact they run Asterisk.

Jeremy McNamara
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Re: [Asterisk-Users] BRI in the US

[Mike Boger Jr]

Brian,

Are you not able to get BRI where you are?

Maybe the better question is what are you trying to acomplish? I've got a
BRI in the house (with the horribly expensive Eicon sh*t that I paid 30
bucks for on ebay)

Mike

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Re: [Asterisk-Users] SysMaster and GPL Violation

[Mark Spencer]

I hope this is the last time I have to comment on this...
I'm struggling to think of another free software project where contributed
code bearing an identical GPL or BSD license would require any such
additional disclaimer.
How about GCC, GLIBC, etc?  Do you think Digium came up with those 
disclaimers on our own?  No, we used the FSF ones with the exception that 
the "long version" of ours has the original author retain copyright with 
Digium as unlimited licensee, whereas the FSF requires copyright 
assignment to the FSF and an unlimited license back to the original 
author.  This distinction is because the FSF wants to be able to enforce 
the GPL on code they did not author, whereas with Digium and Asterisk we 
are more concerned that we do not allow contributions to pollute our 
ability to make licensing decisions.

Again, as with the FSF projects, nobody is *required* to make any such 
disclaimer -- only if you want your changes to be included in what we 
distribute as Asterisk.  By and large, I think people are willing to do so 
because Digium has worked very hard to keep adding value to Asterisk and 
because we've stuck to our guns on being sure that Digium created code 
always remains Open Source and available under GPL.  Personally, I am very 
much aware of the trust the Asterisk community has given to Digium, and 
likewise I am very much aware the security and the independence *from* 
Digium that GPL gives them.  Our future is totally dependent upon our 
ability to continue to earn the support, help and loyalty of the community 
at large.

Mark
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Re: [Asterisk-Users] getting callerid from spa3k to asterisk

[Steve Totaro]

Maybe this will be of help.
http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=1707&PN=0&TPN=1
- Original Message - 
From: "Randy Bush" <[EMAIL PROTECTED]>
To: "splatters" <[EMAIL PROTECTED]>
Sent: Friday, November 12, 2004 9:13 PM
Subject: [Asterisk-Users] getting callerid from spa3k to asterisk


ok, with a good pointer from Chris Stenton <[EMAIL PROTECTED]>,
i found the problem.
if i have two sip contexts for my spa3k, on inbound and
one outbound, e.g.
   [spa3k-out]
   type=peer
   auth=md5
   secret=pfui
   username=outpass
   fromuser=outpass
   host=spa3k.bogus.com
   port=5061
   nat=no
   canreinvite=yes
   context=ext-in42
   [spa3k-in]
   type=friend
   host=dynamic
   port=5061
   auth=md5
   secret=pfui
   qualify=1000
   canreinvite=yes
   context=ext-in42
and the spa3k's PSTN / Subscriber Information / User ID: = spack-in,
the incoming connection from spa3k to * is being routed to the
spa3k-out context, not the spa3-in context.  see appended.
i suspect this is a bug in * 1.0.1.
so, until the problem is diagnosed, how do i work around it.
as the spa3k is registered, i tried to remove the spa3k-out
context entirely.  callerid now works.  yes!
but ...  if i try to place an outbound call using the spa3k-in
context, the call is sent to the spa3k, but it just gives me
the pstn's dialtone, and does not dial the number.  my spa3k
config is in .
so how do i place a call out the spa3k pstn without a separate
outbound context?
randy
---
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1
From: CallerName 
;tag=25aee11517d597a1o1
To: 
Remote-Party-ID: CallerName 
;screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: biwa 0431 
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 428
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 8805171 8805171 IN IP4 198.180.150.195
s=-
c=IN IP4 198.180.150.195
t=0 0
m=audio 16396 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 19 lines
Using latest request as basis request
Sending to 198.180.150.195 : 5061 (non-NAT)
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 198.180.150.195:16396
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - 
audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 
0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
Found peer 'spa3k-out'

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RE: [Asterisk-Users] BRI in the US

[Scott Stingel]

Brian-

I was quoted (verbally) something on the order of $60 per month for a single
BRI by SBC in San Francisco about 60 days ago.  I thought that was high..

Regards
Scott  


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, November 12, 2004 1:23 PM
To: 'Michael Bielicki'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] BRI in the US

Check this out www.bkw.org/pri.pdf

That's what SBC charges for PRI here... it's the only option I have right
now.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of Michael Bielicki
> Sent: Friday, November 12, 2004 2:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BRI in the US
> 
> the horribly expensive EICON shit. Although if you just want to 
> connect ISDN phones to asterisk you can use european ISDN phones with 
> cards from Junghanns.net
> 
> 
> On Fri, 12 Nov 2004 14:10:07 -0600, Brian West <[EMAIL PROTECTED]> wrote:
> > What cards will work with asterisk and BRI in the US?
> >
> > bkw
> >
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> 
> 
> --
> Michael Bielicki
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Re: [Asterisk-Users] Advice on starting out

[Steve Totaro]

yeah, sounds like you have done your homework very well.
- Original Message - 
From: "Jeffrey C Honig" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Friday, November 12, 2004 9:20 PM
Subject: [Asterisk-Users] Advice on starting out


I'm looking to start playing with Asterisk as soon as I get some H/W.
I tele-commute from home and have three POTS lines, home, work and work
FAX.
I'm looking at a setup that will have a couple high end phones (3-6) SIP
phones, a couple medium SIP phones, a couple of inexpensive SIP phones
and a couple of FXS ports for cordless phones (until 802.11 SIP phones
become viable).  I'll probably drop down to two POTS lines and add an
IAX(2) connection to a VoIP provider.  And I may want to add a
doorphone.
I've spent hours perusing the Wiki and have been monitoring this list
for a little while.
At first I was thinking of a TDM400Ps for ports.  But after the recent
posts it looks like a better solution would be a pair of SPA-3000s
(three if I want a door phone) and SPA-2000s for additional FXS ports.
For the medium and high-end phones, the Polycomm IP-500 and IP-600 look
like a good balance of functionality and price.  There seem to be lots
of options at the lower end (BudgeTone, Sipura).
Am I headed in the right direction here?
Thanks.
Jeff
--
Jeffrey C. Honig <[EMAIL PROTECTED]>
http://www.honig.net/jch
GnuPG ID:14E29E13 
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[Asterisk-Users] Advice on starting out

[Jeffrey C Honig]

I'm looking to start playing with Asterisk as soon as I get some H/W.

I tele-commute from home and have three POTS lines, home, work and work
FAX.

I'm looking at a setup that will have a couple high end phones (3-6) SIP
phones, a couple medium SIP phones, a couple of inexpensive SIP phones
and a couple of FXS ports for cordless phones (until 802.11 SIP phones
become viable).  I'll probably drop down to two POTS lines and add an
IAX(2) connection to a VoIP provider.  And I may want to add a
doorphone.

I've spent hours perusing the Wiki and have been monitoring this list
for a little while.

At first I was thinking of a TDM400Ps for ports.  But after the recent
posts it looks like a better solution would be a pair of SPA-3000s
(three if I want a door phone) and SPA-2000s for additional FXS ports.

For the medium and high-end phones, the Polycomm IP-500 and IP-600 look
like a good balance of functionality and price.  There seem to be lots
of options at the lower end (BudgeTone, Sipura).

Am I headed in the right direction here?

Thanks.

Jeff

-- 
Jeffrey C. Honig <[EMAIL PROTECTED]>
http://www.honig.net/jch
GnuPG ID:14E29E13 
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Re: [Asterisk-Users] Calling an outside number along side otherinternal extensions?

[Paul Fielding]

Hmmm... Interesting that you mention it's not a problem with VOIP companies 
as they use PRI.  The analog trunk I'm connecting to is actually a Vonage 
line.  Thing is, it seems to me that by connecting via the Zap channel to 
the Vonage ATA I'm effectively cancelling any advantage that Vonage's PRI 
might have... (?).I don't believe I have any other alternatives for 
connecting to Vonage's service, but perhaps I'm wrong about that.

Perhaps I'll give the "c" option a try.  It looks like it might do the 
job...

regards,
Paul
- Original Message - 
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Friday, November 12, 2004 6:56 PM
Subject: Re: [Asterisk-Users] Calling an outside number along side 
otherinternal extensions?


Paul Fielding wrote:

I've currently configured incoming calls to simultaneously ring an analog 
phone (via TDM400P) and two SIP phones.   I'd like to have it also 
simultaneously dial out the TDM400P on a PSTN to ring my cell phone, and 
have the first one to answer win the battle.

 In my digging I've figured out that I can add the Zap channel to the 
dial list, such as Dial(SIP/7001&SIP/7002&ZAP/3/5551212,20), however when 
I include the PSTN line in this command (ZAP/3/) I get an interesting 
thing happening.

 All SIP phones start ringing.
Asterisk connects ZAP/3 to dial out and dials out
Asterisk then says to the effect of "ZAP/3 has answered the call" (since 
the line has now gone off hook) and stops ringing all the SIP phones 
immediately, leaving me with only the cell phone ringing.  It then fails 
to go to Voicemail and just keeps ringing the cell phone, because as far 
as Asterisk is concerned the line has been bridged and is connected.

 Any suggestions?
Analog FXO ports ae considered "answered" as soon as the dialing is 
finished.  Nothing you can do about this because there is no way for 
Asterisk to know when the far end answers.  This is not a problem with 
(most) Channelized Voice T-1, it's not a problem with PRI and not a 
problem with VoIP telephone companies, since they all use PRI.

You can sort of work around this problem by using the poorly documented 
"c" option to the Zap dial command.  Something like Zap/1c or something 
like that.  I've never used it.  That option requires the callee press # 
to accept the call.  No sound file is played.  See the mailing list 
archives.  It's been discussed off and on.

--Eric
--Eric
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[Asterisk-Users] getting callerid from spa3k to asterisk

[Randy Bush]

ok, with a good pointer from Chris Stenton <[EMAIL PROTECTED]>,
i found the problem.

if i have two sip contexts for my spa3k, on inbound and
one outbound, e.g.

[spa3k-out]
type=peer
auth=md5
secret=pfui
username=outpass
fromuser=outpass
host=spa3k.bogus.com
port=5061
nat=no
canreinvite=yes
context=ext-in42

[spa3k-in]
type=friend
host=dynamic
port=5061
auth=md5
secret=pfui
qualify=1000
canreinvite=yes
context=ext-in42

and the spa3k's PSTN / Subscriber Information / User ID: = spack-in,

the incoming connection from spa3k to * is being routed to the
spa3k-out context, not the spa3-in context.  see appended.

i suspect this is a bug in * 1.0.1.

so, until the problem is diagnosed, how do i work around it.
as the spa3k is registered, i tried to remove the spa3k-out
context entirely.  callerid now works.  yes!

but ...  if i try to place an outbound call using the spa3k-in
context, the call is sent to the spa3k, but it just gives me
the pstn's dialtone, and does not dial the number.  my spa3k
config is in .

so how do i place a call out the spa3k pstn without a separate
outbound context?

randy

---

Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1
From: CallerName  ;tag=25aee11517d597a1o1
To: 
Remote-Party-ID: CallerName  ;screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: biwa 0431 
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 428
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 8805171 8805171 IN IP4 198.180.150.195
s=-
c=IN IP4 198.180.150.195
t=0 0
m=audio 16396 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

15 headers, 19 lines
Using latest request as basis request
Sending to 198.180.150.195 : 5061 (non-NAT)
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 198.180.150.195:16396
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - 
audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 
0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'spa3k-out'

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Re: [Asterisk-Users] OT: Grandstream problems

[Jim Dossey]

On Fri, 2004-11-12 at 16:16 -0600, Rich Adamson wrote:


> > I just got a Grandstream 100 yesterday.  I plugged it into my network and after 
a few 
> seconds it showed the correct
> > time and date.  After that I tried to access the built-in web server.  After 
several 
> tries I finally got the login screen.  That
> > took about 10 tries before it displayed anything.  Then I entered the default 
password 
> 'admin' and clicked the Login
> > button.  After that I got nothing.  The phone still seemed to be working - I 
could press 
> the Menu button and the menu
> > would come up and I could scroll through it.
> > 
> > I also tried changing out the ethernet cable, and plugging it into different 
ports on my 
> Linksys switch (BEFSX41), but
> > nothing works.
> > 
> > Do I just have a bad unit?
> 
> Be carefull with assuptions regarding the Linksys. There were several
> different models (versions) of that hardware, and the early models
> did not support 100 meg ethernet worth a darn. Some packets make
> it through and a large number did not. Change the port speed on
> your phone to 10 meg (if you can) and try again. (I seen your post
> on the ping packet loss and that sort of reminded me of problems
> seen before.)
> 
> Rich
> 
> Thanks for the feedback (and the reply from Dave Cotton as well).  I'm going to try 
the phone on a completely
> different network and see what happens.
> 
> For the record, the problem was an incompatibility between the Budgetone and the Linksys 
ethernet ports.  I plugged a
> KTI 10Mb hub between the Linksys and the Budgetone and it started working.  Don't plug a 
Budgetone 101 into a
> Linksys BEFSX41.

OR, downgrade the Budgetone to 10 meg, or, replace the linksys with a newer one.



How do you downgrade the Budgetone to 10Mb?  I don't see anything on the configuration page to do that.  Also the specs on the Budgetone say it is a 10Base-T port.



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Re: [Asterisk-Users] Calling an outside number along side other internal extensions?

[Eric Wieling]

Paul Fielding wrote:

I've currently configured incoming calls to simultaneously ring an 
analog phone (via TDM400P) and two SIP phones.   I'd like to have it 
also simultaneously dial out the TDM400P on a PSTN to ring my cell 
phone, and have the first one to answer win the battle.

 

In my digging I've figured out that I can add the Zap channel to the 
dial list, such as Dial(SIP/7001&SIP/7002&ZAP/3/5551212,20), however 
when I include the PSTN line in this command (ZAP/3/) I get an 
interesting thing happening.

 

All SIP phones start ringing.
Asterisk connects ZAP/3 to dial out and dials out
Asterisk then says to the effect of "ZAP/3 has answered the call" (since 
the line has now gone off hook) and stops ringing all the SIP phones 
immediately, leaving me with only the cell phone ringing.  It then fails 
to go to Voicemail and just keeps ringing the cell phone, because as far 
as Asterisk is concerned the line has been bridged and is connected.

 

Any suggestions?
Analog FXO ports ae considered "answered" as soon as the dialing is 
finished.  Nothing you can do about this because there is no way for 
Asterisk to know when the far end answers.  This is not a problem with 
(most) Channelized Voice T-1, it's not a problem with PRI and not a 
problem with VoIP telephone companies, since they all use PRI.

You can sort of work around this problem by using the poorly documented 
"c" option to the Zap dial command.  Something like Zap/1c or something 
like that.  I've never used it.  That option requires the callee press # 
to accept the call.  No sound file is played.  See the mailing list 
archives.  It's been discussed off and on.

--Eric
--Eric
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[Asterisk-Users] Calling an outside number along side other internal extensions?

[Paul Fielding]

I've currently configured incoming calls to 
simultaneously ring an analog phone (via TDM400P) and two SIP 
phones.   I'd like to have it also simultaneously dial out the TDM400P 
on a PSTN to ring my cell phone, and have the first one to answer win the 
battle.
 
In my digging I've figured out that I can add the 
Zap channel to the dial list, such as 
Dial(SIP/7001&SIP/7002&ZAP/3/5551212,20), however when I include the 
PSTN line in this command (ZAP/3/) I get an interesting thing 
happening.
 
All SIP phones start ringing.
Asterisk connects ZAP/3 to dial out and dials 
out
Asterisk then says to the effect of "ZAP/3 has 
answered the call" (since the line has now gone off hook) and stops ringing all 
the SIP phones immediately, leaving me with only the cell phone ringing.  
It then fails to go to Voicemail and just keeps ringing the cell phone, because 
as far as Asterisk is concerned the line has been bridged and is 
connected.
 
Any suggestions?
 
regards,
 
Paul
 
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Re: [Asterisk-Users] pressing a key to get out of voicemail?

[Paul Fielding]

Go figure.  I guess I need to use 'show applications' more.  I searched all 
over for docos on the Voicemail app and it was under my nose the whole 
time... :)  Guess I'm on my learning curve.  thanks a bunch...

Paul
- Original Message - 
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Friday, November 12, 2004 6:34 PM
Subject: Re: [Asterisk-Users] pressing a key to get out of voicemail?


Paul Fielding wrote:

I've currently got Asterisk configured to take incoming calls and send 
them directly to my voicemail.  I'd prefer to keep this approach rather 
than sending people to a menu first.

 What I want to be able to do is have voicemail come up, but if someone 
presses a key, such as 9 or 8 or perhaps a combo 98 or such, have it 
break out of voicemail and let me authenticate a password, and upon 
succeeding let me back into a dialplan so I can dial extensions or 
another outside line.
This is from "show application voicemail"
If the caller presses '0' (zero) during the prompt, the call jumps to
extension 'o' in the current context.
If the caller presses '*' during the prompt, the call jumps to
extension 'a' in the current context.
Of course if you press # it will exit out of voicemail to.  Voicemail will 
even TELL you to press # after leacing a message.  Once voicemail exits, 
of course the dialplan will continue at the next priority.
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[Asterisk-Users] Authenticate or DISA?

[Paul Fielding]

I want to authenticate to the phone system, then be 
able to call an extension or dial an outside line.   My preferred 
method would be to use DISA, because a) it's non-verbal - ie. it doesn't talk, 
just provides dialtone, and b) it provides dialtone.
 
However, it seems to be unreliable.  when I 
phone in, sometimes it doesn't seem to recognize my DTMF, and just keeps giving 
a dialtone without authenticating.   It's inconsistent.  I can 
phone the system and it'll work, phone again and it doesn't, phone a third time 
and it works.
 
My alternative seems to be to use Authenticate, and 
upon authenticating simply send the caller to the appropriate context to punch 
in extensions or calls.  The problem with this is a) it voices the 
authentication - ie "please enter password" which to me is inviting people to 
try to figure it out, and b) after authenticating you don't get a dialtone, just 
silence.
 
But at least it works reliably every 
time.
 
Any thoughts?
 
regards,
 
Paul
 
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Re: [Asterisk-Users] pressing a key to get out of voicemail?

[Eric Wieling]

Paul Fielding wrote:

I've currently got Asterisk configured to take incoming calls and send 
them directly to my voicemail.  I'd prefer to keep this approach rather 
than sending people to a menu first.

 

What I want to be able to do is have voicemail come up, but if someone 
presses a key, such as 9 or 8 or perhaps a combo 98 or such, have it 
break out of voicemail and let me authenticate a password, and upon 
succeeding let me back into a dialplan so I can dial extensions or 
another outside line.
This is from "show application voicemail"
If the caller presses '0' (zero) during the prompt, the call jumps to
extension 'o' in the current context.
If the caller presses '*' during the prompt, the call jumps to
extension 'a' in the current context.
Of course if you press # it will exit out of voicemail to.  Voicemail 
will even TELL you to press # after leacing a message.  Once voicemail 
exits, of course the dialplan will continue at the next priority.
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[Asterisk-Users] pressing a key to get out of voicemail?

[Paul Fielding]

I've currently got Asterisk configured to take 
incoming calls and send them directly to my voicemail.  I'd prefer to keep 
this approach rather than sending people to a menu first.
 
What I want to be able to do is have voicemail come 
up, but if someone presses a key, such as 9 or 8 or perhaps a combo 98 or such, 
have it break out of voicemail and let me authenticate a password, and upon 
succeeding let me back into a dialplan so I can dial extensions or another 
outside line.
 
It appears that there's no way to alter the 
Voicemail app behavior?
 
So far the only way I've come up with to do this is 
to cheat.  Instead of going straight to voicemail I've set it to play a wav 
file that Backgrounds "This is my voicemail, leave a message.. yada yada", then 
sends the call to Voicemail, only my Voicemail unavailable message is an empty 
wave file.  This allows me to press another extension while the first wave 
file is being played, and as long as I do it before it jumps into voicemail, I 
can break to another context where I can Authenticate, then send where I 
want.
 
But this is a kludge, and I cannot change my 
voicemail message using regular voicemail tools this way.  I'd rather set 
it up properly.
 
Any ideas?
 
regards,
 
Paul
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Re: [Asterisk-Users] quasi-skype channel for Asterisk?

[James H. Thompson]

I created a wiki page for Skype gateways.
(BSo far the only two I've heard about are the PCphoneonline and the Siemens 
(BGigaset gateway:
(BIf more let me know or add to wiki:
(B
(Bhttp://www.voip-info.org/wiki-Skype+Gateways
(B
(B
(BThanks.
(B
(BJim
(B
(BJames H. Thompson
(B[EMAIL PROTECTED]
(B
(B___
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[Asterisk-Users] voip to pstn

[Tianu Mihai - Cristian]

Is there an example config for asterisk voip to pstn ?

I have the following scenario 

TDM400P with 2xFXO connected to the phone lines + asterisk > internet 
-> TDM400P 2x FXS + asterisk !

So far i have managed to Pick -up a call incoming from pstn to the fxs , but 
it's not working in the other way ! 

Configs :

---> sip.conf

Asterisk 1 (FXO)
[general]
context=from-sip; Default context for incoming calls
port=5060   ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
;srvlookup=yes  ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the 
Internet
;tos=184; Set IP QoS to either a keyword or numeric 
val
;tos=lowdelay   ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing 
registration
notifymimetype=text/plain   ; Allow overriding of mime type in MWI NOTIFY
;videosupport=yes   ; Turn on support for SIP video


allow=all
allow=gsm
allow=g723.1
allow=g729
allow=ulaw

musicclass=default  ; Sets the default music on hold class for all 
SIP calls
; This may also be set for individual 
users/peers
language=en ; Default language setting for all users/peers
; This may also be set for individual 
users/peers
relaxdtmf=yes   ; Relax dtmf handling
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no  ; If we should generate in-band ringing always
useragent=Asterisk PBX  ; Allows you to change the user agent string
nat=no  ; NAT settings
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581

promiscredir=yes
register => [EMAIL PROTECTED]

[21]
type=friend
user=21
fromuser=21
secret=1234
host=dynamic
nat=0
allow=all

[22]
type=friend
user=22
fromuser=22
secret=1234
host=dynamic
nat=0
allow=all

extensions.conf

[globals]



[general]
;
static=yes
writeprotect=yes
;

[extensions]
;
;ton de test
;
exten => 11,1,Milliwatt()
exten => 11,2,Hangup
;
; Data si Timp
;
exten => 13,1,DateTime()
exten => 13,2,Wait(1)
exten => 13,3,DateTime()
exten => 13,4,Hangup
;
exten => 21,1,Dial(SIP/21,20) 
exten => 21,2,Voicemail(u21)
exten => 21,3,Hangup
;
exten => 22,1,Dial(SIP/22,20)   ;
exten => 22,2,Voicemail(u22)
exten => 22,3,Hangup

[linia1-centrala]
exten => s,1,Dial(SIP/21,20)
exten => s,2,Hangup

[linia2-centrala]
exten => s,1,Dial(SIP/22|20)
exten => s,2,Hangup

[default]
include => linia1-centrala
include => linia2-centrala
;
[from-sip]
include => extensions
exten => 0,1,Dial(Zap/g1/${EXTEN},70)
exten => _XX,1,Dial(Zap/g1/${EXTEN},70)
;


the other machine 

sip.conf

[general]
context=from-sip; Default context for incoming calls
port=5060   ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
;srvlookup=yes  ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the 
Internet
;tos=184; Set IP QoS to either a keyword or numeric 
val
;tos=lowdelay   ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing 
registration
notifymimetype=text/plain   ; Allow overriding of mime type in MWI NOTIFY
;videosupport=yes   ; Turn on support for SIP video


allow=all
allow=gsm
allow=g723.1
allow=g729
allow=ulaw

musicclass=default  ; Sets the default music on hold class for all 
SIP calls
; This may also be set for individual 
users/peers
language=en ; Default langu

[Asterisk-Users] Grandstream BT100 - No Sound with Playback()

[davis]

Hi Everyone,

   I'm having a problem with a Grandstream Budge Tone 100 phone.  When
Asterisk send sound to the extension using Playback I'm getting the
following message:

-- Executing Playback("SIP/2002-01fe", "tt-monkeysintro") in new stack
-- Playing 'tt-monkeysintro' (language 'en')
Nov 11 11:54:02 WARNING[278540]: file.c:550 ast_readaudio_callback: Failed
to write frame
  == Spawn extension (from-sip, , 1) exited non-zero on 'SIP/2002-01fe'

I tried every codec on phone with no succes...

Here is the entry for the Grandstream phone in my sip.conf
[2002]
type=friend   ; either "friend" (peer+user), "peer" or
context=from-sip
username=2002 ; usually matches the section title
fromuser=2002 ; overrides the callerid, e.g. required by FWD
secret=123456
callerid=John Doe <2002>
host=dynamic ; we have a static but private IP address
nat=no   ; there is not NAT between phone and Asterisk
;canreinvite=yes   ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc2833  ; either RFC2833 or INFO for the BudgeTone
;[EMAIL PROTECTED]  ; mailbox 1234 in voicemail context "default"
;allow=all  ; need to disallow=all before we can use allow
disallow=all
allow=ulaw

The BT100 and Asterisk are on the same lan...  It's look like every time
the playback function is executed the BT100 just hangup.

Thanks for your help,
Dave
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[Asterisk-Users] Grandstream BT100 - No Sound with Playback()

[davis]

Hi Everyone,

   I'm having a problem with a Grandstream Budge Tone 100 phone.  When
Asterisk send sound to the extension using Playback I'm getting the
following message:

-- Executing Playback("SIP/2002-01fe", "tt-monkeysintro") in new stack
-- Playing 'tt-monkeysintro' (language 'en')
Nov 11 11:54:02 WARNING[278540]: file.c:550 ast_readaudio_callback: Failed
to write frame
  == Spawn extension (from-sip, , 1) exited non-zero on 'SIP/2002-01fe'

I tried every codec on the phone with no succes...

Here is the entry for the Grandstream phone in my sip.conf
[2002]
type=friend   ; either "friend" (peer+user), "peer" or
context=from-sip
username=2002 ; usually matches the section title
fromuser=2002 ; overrides the callerid, e.g. required by FWD
secret=123456
callerid=John Doe <2002>
host=dynamic ; we have a static but private IP address
nat=no   ; there is not NAT between phone and Asterisk
;canreinvite=yes   ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc2833  ; either RFC2833 or INFO for the BudgeTone
;[EMAIL PROTECTED]  ; mailbox 1234 in voicemail context "default"
;allow=all  ; need to disallow=all before we can use allow
disallow=all
allow=ulaw

The BT100 and Asterisk are on the same lan...  It's look like every time
the playback function is executed the BT100 just hangup.

Thanks for your help,
Dave


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[Asterisk-Users] DECT channel

[Chris Ostler]

I have a DECT cordless phone that I've been using just as a standalone
phone.  The interesting thing is that the phone (CyberGenie) was developed
to run with Windows PBX software.  I've been looking at the possibilities of
getting it to work with Asterisk.  Although the obvious solution would be to
get an FSX card and plug the phone into it, I tend to think (or should that
be dream) bigger.

The base station contains a PTSN line in and a PTSN line out in addition to
the DECT base station.  It connects to a PC via a USB connection, providing
three device endpoints.  What I would *really* like to do is to get the line
in working as a FSO, the line out working as a FSO, and the DECT base
station providing a channel for each handset.

I've seen references on the list to the COM-ON-AIR line of DECT products.
Has anyone been successful in getting them to work with Asterisk?  If so,
what was involved in doing so?

The one benefit I have is that I have access to code for the Windows device
drivers and application, and so should be able to use that as a reference.
Although I don't have too much time to give to it, I would also be able to
work to code/adapt needed drivers.

Chris

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RE: [Asterisk-Users] Need low-cost flat-rate incoming DID's throughoutthe U.S - Anybody competing against VoicePulse?

[Jean-François Rousseau]

Could people reply to the list too ? I would be interested to know about
this too.

Thanks 


___
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Sys-Tech
www.sys-tech.net
[EMAIL PROTECTED]
Tél. 24h (418) 520-0739

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Paul Rodan
Envoyé : 12 novembre 2004 18:04
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : [Asterisk-Users] Need low-cost flat-rate incoming DID's
throughoutthe U.S - Anybody competing against VoicePulse?

Is there a company out there that can offer flat rate DID's in 75% or more
of the U.S? Or at least in every major populated area? Right now we use
VoicePulse but they're way overpriced per DID and we've had quality issues.
Not only that, they can't port. We need somebody that can port numbers over
as well. Too many times we've had a sale in another city and we've lost it
because we couldn't port their existing numbers or we charged too much for
each DID (thanks to VoicePulse).

There has to be a company out there that can do something very similar but
at least at half the cost. 

For National calling we're aware of:
LookieLoo = $0.0065 a minute on-network and $0.01 off-network VoiceJet =
$0.013 a minute NuFone = $0.02 a minute VoicePulse = $0.0295 a minute

However the only one that can get us unlimited National did's are
voicepulse. LookieLoo can get us National DID's but at a per minute incoming
rate. NuFone can only do Michigan and I don't know about VoiceJet.

Please reply to me off list as this forum is just way too active for me for
me to read everything.


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[Asterisk-Users] CNG Comfort Noise Generation

[Assaf Benharoosh]

I have a problem 
with many phone such as BudgeTone, ariaVoice, PCPhoneline. They are not 
generating comfort noise (you can hear yourself when you're talking)- with 
budgetone having CNG sporadically.
 
Is there a way to 
make this happen on Asterisk - or it must be a phone 
feature.
 
Does anyone else 
experiencing this issue with those phones and have a 
workaround?
 
Assaf Benharoosh
 
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Re: [Asterisk-Users] Configuring Asterisk As A Sip Server

[Sean Hull]

> I want to configure my Asterisk Server As a SIP is there any 
> possibality.How i do that.Any help is highly appreciated.

Adnan:

I was exactly where you are when I started.  I wrote a doc which might 
help you:
http://iheavy.com/modules.php?op=modload&name=News&file=article&sid=35&mode=thread&order=0&thold=0


Sean

-- 
Sean Hull, Senior Consultant
Heavyweight Internet Group
Rockefeller Center, Box 5352
New York, NY 10185
http://www.iheavy.com 
voice: 646.827.9877x23 fax: 646.827.3434

Sean Hull, founder and senior consultant of Heavyweight Internet Group
is the author of O'Reilly and Associates "Oracle and Open Source"
bridging Open Source software and integration with the world's best
performing database, Oracle. http://www.oreilly.com/catalog/oracleopen/  

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[Asterisk-Users] Newbie question

[Julio Tejera]

 Hello:
 
 First, I'm really new to asterisk and I'm testing it in order
to improve my first steps...
 
 Recently I installed * asterisk on a FreeBSD Box (5.2.1)
I got it working on my internal LAN (it works fine !).
 
I was trying to connect my * box through FWD using SIP
but it is not working an I'm very confused about *, in fact
I can't call from my * client (X-Lite) to a FWD number,
but bettwen my * sip authenticated clients yes...

Please somebody can help me or guide me to the right direction ?

Any kind of help will be appreciated and excuseme by my english :o(

Here is attached my config files
 




extensions.conf
Description: Binary data


sip.conf
Description: Binary data
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Re: [Asterisk-Users] SysMaster and GPL Violation

[James Taylor]

I apologise.
It was not right for me to reject Joe's argument based on some irrelevant  
fact about him.  And now that I anaylize my remarks, it could have been  
misunderstood as a personal attack.

Obviously I had a definite error in my reasoning
Once again, Joe, please accept my apology.
Can we establish a "GPL Viloation" list somewhere so this can be  
continuted there?
I'd really like to get back to reading about functional problems with  
running the Asterisk software.

James
P.S. Brian,
Thanks for my refresher course in Logical Fallacies.
On Fri, 12 Nov 2004 16:31:14 -0600, Brian Capouch <[EMAIL PROTECTED]>  
wrote:

James Taylor wrote:


 I see that your TLD is SOL.net.
I guess that says it all...
Gosh, folks, no ad hominems, PLEASE.
One of the things I like most about this list is that things rarely  
devolve to that level.

I get mad when I read Joe's posts, but at the content, not the person. I  
suspect he's a pretty good person overall, like we all are :-)

B.
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--
Using Opera's revolutionary e-mail client: http://www.opera.com/m2/
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RE: [Asterisk-Users] SysMaster and GPL Violation

[Paul Rodan]

Damn it. Sorry guys (and girls), this wasn't supposed to go here. 

I accidentally selected the wrong list from the drop down outlook presented
me when I began to type "asterisk-"

This is commercial related and I have sent it to the asterisk-biz list.
Sorry for the inconvenience. However, if anybody has an answer to the
question below, I'm interested.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan
Sent: Friday, November 12, 2004 6:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SysMaster and GPL Violation

I concur. This thread needs to end. 

I think Sysmaster's smartest move is to pay for the commercial license; but
I can wish that they'd just release the source code to their very impressive
web interface and other additions :-) 

It doesn't matter what any of us think though, it's Digium's software, it's
their decision on how they wish to proceed. I'm sure by the length of this
thread, they're aware of it and have already made a decision on their next
course of action.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, November 12, 2004 5:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SysMaster and GPL Violation

Ok Guys... I started this whole thread in hopes of one of two things.

1. Hope that SysMaster does the right thing and pays digium the boat load of
money they are entitled to.

2. Or comply with the GPL.

In my eyes either way is a good thing.  I personally do not care about the
disclaimer at all it's the glue that holds the puzzle together.   If my work
is in CVS and Digium sells it big woop... I just hope that company does the
right thing and pays for that right.  My goal was not to piss anyone off or
cause anyone any troubles.   If I were someone that paid for SysMaster
thinking it was the cats meow then finds out its just Asterisk with some
mods under the hood I would be pretty pissed because those boxes are 10-45k
(Depending on what you want)  at that rate they can afford to pay the
license fees.   

I think that's the only fair thing to do...  I don't speak for Digium and if
I implied that then I'm sorry I'm just a user and a developer that thinks
this is unacceptable behavior for any company (ie SysMaster).

bkw



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[Asterisk-Users] Need low-cost flat-rate incoming DID's throughout the U.S - Anybody competing against VoicePulse?

[Paul Rodan]

Is there a company out there that can offer flat rate DID's in 75% or more
of the U.S? Or at least in every major populated area? Right now we use
VoicePulse but they're way overpriced per DID and we've had quality issues.
Not only that, they can't port. We need somebody that can port numbers over
as well. Too many times we've had a sale in another city and we've lost it
because we couldn't port their existing numbers or we charged too much for
each DID (thanks to VoicePulse).

There has to be a company out there that can do something very similar but
at least at half the cost. 

For National calling we're aware of:
LookieLoo = $0.0065 a minute on-network and $0.01 off-network
VoiceJet = $0.013 a minute
NuFone = $0.02 a minute
VoicePulse = $0.0295 a minute

However the only one that can get us unlimited National did's are
voicepulse. LookieLoo can get us National DID's but at a per minute incoming
rate. NuFone can only do Michigan and I don't know about VoiceJet.

Please reply to me off list as this forum is just way too active for me for
me to read everything.


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RE: [Asterisk-Users] SysMaster and GPL Violation

[Paul Rodan]

I concur. This thread needs to end. 

I think Sysmaster's smartest move is to pay for the commercial license; but
I can wish that they'd just release the source code to their very impressive
web interface and other additions :-) 

It doesn't matter what any of us think though, it's Digium's software, it's
their decision on how they wish to proceed. I'm sure by the length of this
thread, they're aware of it and have already made a decision on their next
course of action.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday, November 12, 2004 5:45 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SysMaster and GPL Violation

Ok Guys... I started this whole thread in hopes of one of two things.

1. Hope that SysMaster does the right thing and pays digium the boat load of
money they are entitled to.

2. Or comply with the GPL.

In my eyes either way is a good thing.  I personally do not care about the
disclaimer at all it's the glue that holds the puzzle together.   If my work
is in CVS and Digium sells it big woop... I just hope that company does the
right thing and pays for that right.  My goal was not to piss anyone off or
cause anyone any troubles.   If I were someone that paid for SysMaster
thinking it was the cats meow then finds out its just Asterisk with some
mods under the hood I would be pretty pissed because those boxes are 10-45k
(Depending on what you want)  at that rate they can afford to pay the
license fees.   

I think that's the only fair thing to do...  I don't speak for Digium and if
I implied that then I'm sorry I'm just a user and a developer that thinks
this is unacceptable behavior for any company (ie SysMaster).

bkw



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[Asterisk-Users] AgentCallBackLogin

[Shawn Dillon]

I have the AgentCallBackLogin working well when the support
technician logs into the queue manually. If there a way to get certain
extensions to automatically log into the queue? That way I do not have to worry
about help desk staff forgetting to log into the support queue and never
receiving support calls.

 

Thanks

Shawn Dillon






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RE: [Asterisk-Users] SysMaster and GPL Violation

[Brian West]

Ok Guys... I started this whole thread in hopes of one of two things.

1. Hope that SysMaster does the right thing and pays digium the boat load of
money they are entitled to.

2. Or comply with the GPL.

In my eyes either way is a good thing.  I personally do not care about the
disclaimer at all it's the glue that holds the puzzle together.   If my work
is in CVS and Digium sells it big woop... I just hope that company does the
right thing and pays for that right.  My goal was not to piss anyone off or
cause anyone any troubles.   If I were someone that paid for SysMaster
thinking it was the cats meow then finds out its just Asterisk with some
mods under the hood I would be pretty pissed because those boxes are 10-45k
(Depending on what you want)  at that rate they can afford to pay the
license fees.   

I think that's the only fair thing to do...  I don't speak for Digium and if
I implied that then I'm sorry I'm just a user and a developer that thinks
this is unacceptable behavior for any company (ie SysMaster).

bkw



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Re: [Asterisk-Users] SysMaster and GPL Violation

[Michael Loftis]

--On Friday, November 12, 2004 15:25 +0200 Gilad Ben-Yossef 
<[EMAIL PROTECTED]> wrote:

In short, first find out what Mark wants to do with this and let Digium
lawyers go through with this or otherwise you're risking making it more
difficult for Mark & Digium.
Only if Mark is not interested in going the legal path should we make
this Slashdot material.
I couldn't agree more.  Yes we're the community, but the copyright holders 
need to get involved first, and anything we do in that regard should be 
based on their input, not our own.
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RE: [Asterisk-Users] SysMaster and GPL Violation -- what about IPEYA

[Michael Loftis]

--On Friday, November 12, 2004 10:48 -0600 Brian West <[EMAIL PROTECTED]> wrote:
... Granted I have
never been scuba diving so I can't compare it... but I would think that
the air in a scuba tank is better.
Actually it can be worse, depends on who compresses it.
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> >Joe, I think everyone appreciates the fact that you don't like the way 
> >Digium handles the licensing and disclaimers.
> 
> Brian, haven't you guessed by now that Joe is crusade-happy?  Just look at 
> the "top posting" thread and gaze in awe upon the sheer mass of his posts 
> on the subject.  If I were a professional psychiatrist I would have 
> additional comments...

You would, huh.

Sorry to disappoint you, but I'm not crusade-happy.  I am, however, a bit
exasperated by a user community that plays fast and loose with GPL advocacy
when it suits them, and then jumps to Digium's defense when it is pointed
out that the license isn't being applied to the standards desired by the
FSF.

Of course, if you don't want to discuss things rationally, just say so.

Personally, I believe licensing is important for numerous reasons.  To get
back to the original topic, if SysMaster is violating the GPL, then by all
means, they ought to be taken to the cleaners.  I may have some
philosophical disagreements with the GPL, and would probably be highly
unlikely to license any code I wrote under the GPL, but I'll defend your
right to license code under any license you so wish (even if it's "You may
only use this code if your first name begins with X"), and I'm astounded 
that I'm being jumped all over for defending the principles behind the GPL
in view of the Digium licensing model.

It's gotta be a full moon.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Brian Capouch]

James Taylor wrote:


I see that your TLD is SOL.net.
I guess that says it all...
Gosh, folks, no ad hominems, PLEASE.
One of the things I like most about this list is that things rarely 
devolve to that level.

I get mad when I read Joe's posts, but at the content, not the person. I 
suspect he's a pretty good person overall, like we all are :-)

B.
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> The proper forum for your "demand" is to send a certified letter to  
> Digium's agent for service.
> If you don't like the software, don't use it, don't agree to the terms.
> 
> I see that your TLD is SOL.net.
> I guess that says it all...

I see that you're a top-posting idiot who cannot follow attributions to
see that I demanded nothing at all, and simply responded to someone who
did.

Looks like you're SOL on that cheap attack, dude.

Best,

... JG

> On Fri, 12 Nov 2004 22:18:53 +, Martin List-Petersen  
> <[EMAIL PROTECTED]> wrote:
> 
> > Citat Joe Greco <[EMAIL PROTECTED]>:
> >
> >> > I too demand sysmaster either pay Digium for a non-gpl license or
> >>
> >> Now, here, this gets to the heart of a problem I've hinted at before.
> >>
> >> Digium is making people sign a draconian agreement that gives up rights
> >> to patches and features that are integrated into Asterisk, by signing
> >> rights over to Digium.
> >
> > Are you sure, that you have read the disclaimers ? There are two ways:  
> > you make
> > the software public domain, giving up the right, but not giving them to  
> > anybody
> > OR you give digium a non-execlusive (read NON), non-revocable right to  
> > use your
> > modifications. The copyright stays with you.
> >
> > Please go back and read the disclaimer before making such statements.
> >
> > Kind regards,
> > Martin List-Petersen
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
> 
> 
> -- 
> Using Opera's revolutionary e-mail client: http://www.opera.com/m2/
> ___
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-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] SysMaster and GPL Violation

[James Taylor]

The proper forum for your "demand" is to send a certified letter to  
Digium's agent for service.
If you don't like the software, don't use it, don't agree to the terms.

I see that your TLD is SOL.net.
I guess that says it all...

On Fri, 12 Nov 2004 22:18:53 +, Martin List-Petersen  
<[EMAIL PROTECTED]> wrote:

Citat Joe Greco <[EMAIL PROTECTED]>:
> I too demand sysmaster either pay Digium for a non-gpl license or
Now, here, this gets to the heart of a problem I've hinted at before.
Digium is making people sign a draconian agreement that gives up rights
to patches and features that are integrated into Asterisk, by signing
rights over to Digium.
Are you sure, that you have read the disclaimers ? There are two ways:  
you make
the software public domain, giving up the right, but not giving them to  
anybody
OR you give digium a non-execlusive (read NON), non-revocable right to  
use your
modifications. The copyright stays with you.

Please go back and read the disclaimer before making such statements.
Kind regards,
Martin List-Petersen
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Tom Lahti]

Joe, I think everyone appreciates the fact that you don't like the way 
Digium handles the licensing and disclaimers.
Brian, haven't you guessed by now that Joe is crusade-happy?  Just look at 
the "top posting" thread and gaze in awe upon the sheer mass of his posts 
on the subject.  If I were a professional psychiatrist I would have 
additional comments...

[snip]
it allows them the freedom to do side things with Asterisk--yes, with my 
code and your code and everyone else's code too--in order to make a 
living, and validate the thought that doing Asterisk as Open Source could 
be done in a way that Digium could still make some money on it.
It is in fact a trade: Brian etc. contribute code and sign it over, and in 
return everyone gets to use Mark's product free of monetary charge.

IMO we are perched at the very beginning of the revolution that Asterisk 
will bring to telephony.  I don't mind watching the arguments made about 
the minutiae of the GPL; I *really* mind seeing Mark's and Digium's 
motives impugned.
If Digium's main purpose for existing is not "to make money", then why on 
earth would Mark create the company in the first place?  It's a hellofa lot 
of work to do for kicks. If anyone were that self-sacrificial he probably 
would have already committed suicide to avoid the possibility of anyone 
suffocating because of the oxygen he himself breathed, long before he 
started a company to (gasp!) make money.

--
Tom
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Martin List-Petersen]

Citat Joe Greco <[EMAIL PROTECTED]>:

> > I too demand sysmaster either pay Digium for a non-gpl license or 
> 
> Now, here, this gets to the heart of a problem I've hinted at before.
> 
> Digium is making people sign a draconian agreement that gives up rights
> to patches and features that are integrated into Asterisk, by signing
> rights over to Digium.

Are you sure, that you have read the disclaimers ? There are two ways: you make
the software public domain, giving up the right, but not giving them to anybody
OR you give digium a non-execlusive (read NON), non-revocable right to use your
modifications. The copyright stays with you.

Please go back and read the disclaimer before making such statements.

Kind regards,
Martin List-Petersen
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Re: [Asterisk-Users] OT: Grandstream problems

[Rich Adamson]

> > I just got a Grandstream 100 yesterday.  I plugged it into my 
> network and after 
a few 
> seconds it showed the correct
> > time and date.  After that I tried to access the built-in web 
> server.  After 
several 
> tries I finally got the login screen.  That
> > took about 10 tries before it displayed anything.  Then I entered 
> the default 
password 
> 'admin' and clicked the Login
> > button.  After that I got nothing.  The phone still seemed to be 
> working - I 
could press 
> the Menu button and the menu
> > would come up and I could scroll through it.
> > 
> > I also tried changing out the ethernet cable, and plugging it into 
> different 
ports on my 
> Linksys switch (BEFSX41), but
> > nothing works.
> > 
> > Do I just have a bad unit?
> 
> Be carefull with assuptions regarding the Linksys. There were several
> different models (versions) of that hardware, and the early models
> did not support 100 meg ethernet worth a darn. Some packets make
> it through and a large number did not. Change the port speed on
> your phone to 10 meg (if you can) and try again. (I seen your post
> on the ping packet loss and that sort of reminded me of problems
> seen before.)
> 
> Rich
> 
> Thanks for the feedback (and the reply from Dave Cotton as well).  I'm 
> going to try 
the phone on a completely
> different network and see what happens.
> 
> For the record, the problem was an incompatibility between the Budgetone and 
> the Linksys 
ethernet ports.  I plugged a
> KTI 10Mb hub between the Linksys and the Budgetone and it started working.  
> Don't plug a 
Budgetone 101 into a
> Linksys BEFSX41.

OR, downgrade the Budgetone to 10 meg, or, replace the linksys with a newer one.


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Re: [Asterisk-Users] OT: Grandstream problems

[Jim Dossey]

On Fri, 2004-11-12 at 12:45 -0500, Jim Dossey wrote:

On Fri, 2004-11-12 at 10:55 -0600, Rich Adamson wrote: 


> I just got a Grandstream 100 yesterday.  I plugged it into my network and after a few 
seconds it showed the correct
> time and date.  After that I tried to access the built-in web server.  After several 
tries I finally got the login screen.  That
> took about 10 tries before it displayed anything.  Then I entered the default password 
'admin' and clicked the Login
> button.  After that I got nothing.  The phone still seemed to be working - I could press 
the Menu button and the menu
> would come up and I could scroll through it.
> 
> I also tried changing out the ethernet cable, and plugging it into different ports on my 
Linksys switch (BEFSX41), but
> nothing works.
> 
> Do I just have a bad unit?

Be carefull with assuptions regarding the Linksys. There were several
different models (versions) of that hardware, and the early models
did not support 100 meg ethernet worth a darn. Some packets make
it through and a large number did not. Change the port speed on
your phone to 10 meg (if you can) and try again. (I seen your post
on the ping packet loss and that sort of reminded me of problems
seen before.)

Rich



Thanks for the feedback (and the reply from Dave Cotton as well).  I'm going to try the phone on a completely different network and see what happens. 


For the record, the problem was an incompatibility between the Budgetone and the Linksys ethernet ports.  I plugged a KTI 10Mb hub between the Linksys and the Budgetone and it started working.  Don't plug a Budgetone 101 into a Linksys BEFSX41.



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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> Joe, I think everyone appreciates the fact that you don't like the way 
> Digium handles the licensing and disclaimers.
> [...] 
> IMO we are perched at the very beginning of the revolution that Asterisk 
> will bring to telephony.  I don't mind watching the arguments made about 
> the minutiae of the GPL; I *really* mind seeing Mark's and Digium's 
> motives impugned.

[snipped down to the heart of it all]

It's not about their motives.  I have no problem seeing Digium make money
off Asterisk.  Heck, we operate a source repository here that a number of 
people probably make money off of (and since it's under a BSD-like license,
no disclaimers are necessary or requested).

What bothers me is that the disclosure of Digium's intent requires a fair
amount of reading in between the lines.  It's a lot less underhanded to
take that disclaimer text and add on:

"Digium reseves the right to release this code under a non-GPL closed
source license.  Do not agree to this disclaimer unless you approve of
this and understand that you are granting them the right to do this."

than it is to have everyone going on and on about "Oh, yeah, this is a GPL
project" without disclosing that in fact there is a scheme to multiply
license the resulting code, by making contributors sign away their rights
(and therefore the protections that would otherwise exist for that code
under the GPL).

I'm all for revolutions in telephony, Lord knows the ILEC's need a kick
in the can.  I'm also all for free software, and I'm happy to see that
Digium has GPL'd Asterisk, but I also believe that there needs to be full
disclosure of the particulars, because what is happening is at conflict
with some of the ideologies behind the GPL.

It's very interesting to see this very list which was so GPL-advocate a
month or two ago turn tail in Digium's defense.  :-)

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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[Asterisk-Users] Kirk IP 600 DECT station

[Remco Barende]

OK, I just got the Kirk IP 600 kit :)
It turns out that they actually make one unit that does H323 and can do 
Skinny (Cisco) if you buy the version with a license for it. Mine supports 
both.

The box appears to be running linux btw :)
nmap revealed 2 open ports, telnet and http and also
TCP/IP fingerprint:
SInfo(V=3.55%P=i686-pc-linux-gnu%D=11/12%Time=419509CD%O=23%C=1)
I couldn't find any info on this station and how to connect it to * but 
that's ok :)

It seems that you must configure the phones as Cisco 7940 in Call Manager. 
The Wiki about the 7940 uses SIP which will not work so I have to try 
skinny or H323.

I think the best way would be to use the skinny protocol but I'm a bit 
lost there. When looking for info on cisco protocol I actually found 3 
channels : chan_skinny / chan_sccp and asterisk-sccp.

Hopefully chan_skinny which comes with * is ok?
The wiki SCCP-HOWTO2 says :
noload=chan_skinny.so
but I guess I do want to load skinny?? should I specify load= ?
I will keep posting results to the list and finally add a wiki when it's 
working, hope any of the cisco experts can help :)

Thanks for any input!
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Re: [Asterisk-Users] Polycom 500 software?

[Chris Tooley]

There are several models of each phone.  The way it breaks down is:

2200-11X00-001 where X is either 3, 5, or 6 (300, 500, 600) is shipped
with a bootrom and no firmware.
2200-11X30-001 where X is either 3, 5 or 6 is shipped with a bootrom
and the SIP firmware
2200-11X40-001 where X is either 3, 5 or 6 is shipped with a bootrom
and the MGCP firmware

The 500 is the only model for which H.323 works and it is only
available through a provider that provides a platform that Polycom has
certified the H.323 firmware with.

If you get a 2200-11530-001 that has SIP Polycom says it cannot be
converted to MGCP.  This appears to be the case in my attempts though
I'm not able to 100% confirm that.


On Mon, 8 Nov 2004 07:29:01 -0700, Damon Estep
<[EMAIL PROTECTED]> wrote:
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Peter Johnson
> > Sent: Monday, November 08, 2004 1:02 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Polycom 500 software?
> >
> > > >
> > > > Polycom ships out two different phones , ones with H323,and one
> with
> > > SIP
> > > > already loaded.
> > > >
> > > > Thank you,
> > > > Steve Maroney
> > >
> > > Correction, the polycom IP 500 ships without h.323 or SIP
> > > software (it only has a bootrom on it), and software is only
> > > distributed by polycom authorized VoIP partners. I have
> > > personally taken issue with this as they advertise the
> > > product as H.323 and SIP compliant, yet without additional
> > > software it does not even know what SIP or H.323 are.
> > >
> >
> > That's not right.
> > New phones come loaded with the current relevant firmware.
> > Upgraded f/w is only available to/from certified resellers.
> > Or look on the wiki for where it is freely available.
> > 
> > Peter
> >
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> 
> 
> Are you sure your statement applies to the IP 500, the two I just RMA'd
> 30 days ago also did not come with any protocol application. At that
> time Polycom confirmed that they do not ship that model with a protocol
> application. With the IP 500 there is a difference between the firmware,
> boot loader, and application. Is it possible in you experience that the
> dealer or distributor loaded the firmware before shipping?
> 
> The normal process for the IP 500 is to download the application you
> want to run (h.323 - aka HMX, SIP, or MCGP) to the phone from a user
> configured FTP server at boot time.
> 
> If polycom has changed this that would be useful to know for sure.
> 
> 
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Brian Capouch]

Michael Giagnocavo wrote:
Again, let Stallman call out nasal demons against whoever he wants.
Meanwhile, I think Digium should be able to do whatever they want with code
they've made, or code disclaimed to them.

Joe, I think everyone appreciates the fact that you don't like the way 
Digium handles the licensing and disclaimers.

I would like to see you say out loud, just once, that those of us who 
know all of that and disclaim our work to Digium are not necessarily 
idiotic boobs who don't know what we're doing.

Digium took a huge risk in making their primary intellectual property 
Open Source.  They didn't have to; this revolutionary system may or may 
not have taken off like it has, had it been proprietary.  But at the 
very least there is a reasonable chance that it would have succeeded as 
such and made them a lot of money.

It tires me to read your tirades and consider that you really don't 
think the rest of us are smart enough to catch Digium's "trick" that 
bothers you so badly.

Things are set up the way they are, as I understand it, so that first, 
Mark gets the final say, totally, as to what is "true Asterisk" and what 
is not.  Second, it allows them the freedom to do side things with 
Asterisk--yes, with my code and your code and everyone else's code 
too--in order to make a living, and validate the thought that doing 
Asterisk as Open Source could be done in a way that Digium could still 
make some money on it.

IMO we are perched at the very beginning of the revolution that Asterisk 
will bring to telephony.  I don't mind watching the arguments made about 
the minutiae of the GPL; I *really* mind seeing Mark's and Digium's 
motives impugned.

B.
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Re: [Asterisk-Users] Quick call group question...

[Chris TenHarmsel]

It's kind of a pain how all the phones show missed calls if they're
not the ones that picked it up though...but I guess that's a phone
issue...I guess.

-Chris


On Fri, 12 Nov 2004 15:23:38 -0500, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> 
> 
> 
> 
> > On Fri, 2004-12-11 at 12:01 -0800, Sean Kennedy wrote:
> >
> >> I remember the syntax is something like this: dial(
> >> SIP/1&SIP/2&SIP/3...yada yada yada ), but what I want to know is this:
> >> If someone answers the extension, does it belong to them at that point?
> >> The other lines will stop ringing, and the other people couldn't pick up
> >> the line?
> >
> > That is correct.  If calling multiple SIP extensions with Dial(), the
> > first extension to answer the call gets it.  Other extensions will stop
> > ringing.
> >
> > Ryan
> 
> This also applies to other channels such as IAX or Zap, if used.
> 
> 
> 
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Re: [Asterisk-Users] BRI in the US

[Joe Greco]

> Check this out www.bkw.org/pri.pdf
> 
> That's what SBC charges for PRI here... it's the only option I have right
> now.

Woww.

I would say they're smokin' the crack.

Are you sure they're the only option?  There are a lot of CLEC's out in
the US...

I know a number of them that used to offer BRI no longer do, so if you
are unable to obtain BRI via Sucky Bell Company, yeah, you may well not
have a lot of other good options.  (delete rant about how SBC has made
BRI a noncompetitive offering in order to force people into PRI...)

It's that sort of pricing which will ultimately drive a lot of stuff to
VoIP.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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[Asterisk-Users] gold rush?

[Tony Nichols]

Another new article with asterisk/Digium in mind

http://www.onlamp.com/pub/wlg/5909

t o n y

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Re: [Asterisk-Users] SIP & ALERT_INFO for distinctive ring

[Cirelle Enterprises]

>- Original Message - 
>From: "Kubat, Philip" <[EMAIL PROTECTED]>
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
><[EMAIL PROTECTED]>
>Sent: Friday, November 12, 2004 1:08 PM
>Subject: RE: [Asterisk-Users] SIP & ALERT_INFO for distinctive ring
>

>Agreed, works fine on creating calls, e.g outbound.
>
>What I am looking for; is there means to capture this (maybe via a variable)
>on call into Asterisk?

This may or may not work for you but this is what we have:

Scenario:

4 inbound lines using a TDM 4 FXO and 4 SIP extensions.

when a call comes in on line 1 ( give a standard ring),
line2 (ring2) , and line3 etc...


[zaptel.conf]
span=1,1,0,esf,b8zs,yellow
nethdlc=1-24

fxsks=25-28<<< 4 pots lines 
  
loadzone = us
defaultzone=us


[zapata.conf]
;fxsks=25
callerid=asreceived
context => incoming-line1
musiconhold = line1moh
signalling=fxs_ks 
channel => 25

;fxsks=26
callerid=asreceived
context => incoming-line2
musiconhold = line2moh
signalling=fxs_ks
channel => 26

;fxsks=27
callerid=asreceived
context => incoming-line3
musiconhold = line3moh
signalling=fxs_ks
channel => 27

;fxsks=28
callerid=asreceived
context => incoming-line4
musiconhold = line4moh
signalling=fxs_ks
channel => 28

loadzone=us
defaultzone=us


[extensions.conf]
...
[incoming-line1]
exten => s,1,NoOp() 
exten => s,2,zapateller()
exten => s,3,SetGlobalVar(CALLSOURCE=121)
exten => s,4,Goto(${CALLSOURCE},${CALLSOURCE},1)
exten => s,5,Hangup

[incoming-line2]
exten => s,1,NoOp() 
exten => s,2,zapateller()
exten => s,3,SetGlobalVar(CALLSOURCE=122)
exten => s,4,Goto(${CALLSOURCE},${CALLSOURCE},1)
exten => s,5,Hangup

(repeat for lines 3 and 4)
...

[121]
exten => 121,1,SetVar(Ext=121)
exten => 121,2,SetVar(Ext2=122)
exten => 121,3,SetVar(ALERT_INFO=Bellcore-r2) < set the ring for calls from 
FXO- line1
exten => 121,4,SetMusicOnHold,homeline 
exten => 121,5,Dial(SIP/${Ext}&SIP/${Ext2},15,Tr)
exten => 121,6,goto(s-${DIALSTATUS},1 )
; go here for no anwer
exten => s-NOANSWER,1,Goto(s-${DIALSTATUS},2)
exten => s-NOANSWER,2,Wait(3)
exten => s-NOANSWER,4,Goto(s-${DIALSTATUS},5)
exten => s-NOANSWER,5,Voicemail(${Ext})
exten => s-NOANSWER,6,Goto(${${Ext}.9)
;go here for a busy line
exten => s-BUSY,1,Goto(s-BUSY,2)
exten => s-BUSY,2,Wait(3)
exten => s-BUSY,4,Goto(s-BUSY,5)
exten => s-BUSY,5,Voicemail(${Ext})
exten => s-BUSY,6,Goto(40852,9)
;done
exten => 40852,9,Hangup

etc... etc... etc... for each extension.


What this does for us is to separate the rings coming from specific pots lines
could also be adapted to other lines (iax, etc...) I believe.

Hope this helps

Greg

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RE: [Asterisk-Users] SysMaster and GPL Violation

[Michael Giagnocavo]

>Well, see what I said.  If you want to be a contributor to this allegedly
>"free software" project, you are forced to sign away your rights.
>
>I'm struggling to think of another free software project where contributed
>code bearing an identical GPL or BSD license would require any such
>additional disclaimer.

You only have to do that if you want Digium to include it in their hosted
version of the Asterisk codebase. You can contribute 'til your heart is
content without signing anything away. It might be quite different than
other projects, but it's not like they are limiting the GPL or anything.

Actually, I'm overjoyed to see them doing this. I *want* Digium to make
money off of Asterisk.

I want them to hire the brightest devs in the world, create BVTs (so it's
not a "checkin and see if it works" kinda thing), etc. etc. etc.  I want
them to have a huge marketing force that gives me all sorts of awesome
materials to sell Asterisk  -- ever gotten one of those MS sales kits? Some
of them even have fully scripted sales pitches, inc. jokes and so on. 

Let's face it, if Digium was spending $1million a month on R&D, it'd benefit
all of us (assuming they could afford it and it didn't drive 'em into the
ground :P).

BTW, this isn't a jab at OSS (any replies similar to "j00 want to see digium
become M$ so you sux0rs" will be ignored). I could care less what license a
product is under, as long as it works for me. I want the best software, and
unlike Stallman, I think people have every bloody right to do whatever they
want with what they create, inc. earning billions of dollars if they can do
so.

-Michael


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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> On Fri, 2004-11-12 at 12:24 -0600, Joe Greco wrote:
> > > On Fri, 2004-11-12 at 09:26 -0600, Joe Greco wrote:
> > > > > I too demand sysmaster either pay Digium for a non-gpl license or 
> > > > 
> > > > Now, here, this gets to the heart of a problem I've hinted at before.
> > > > 
> > > > Digium is making people sign a draconian agreement that gives up rights
> > > > to patches and features that are integrated into Asterisk, by signing
> > > > rights over to Digium.
> > > > 
> > > > I would expect that most contributors do not realize that they are 
> > > > setting
> > > > up a scenario where Digium can, in fact, sell non-GPL Asterisk licenses 
> > > > to 
> > > > third parties and essentially sell their work.
> > > 
> > > I think we all knew that. In fact, we consider it a good thing as it
> > > allows Digium an income to keep paid developers working on the code
> > > base.
> > 
> > Really?  Wouldn't it be nice, then, if Digium explicitly stated that this
> > was their intention, in their little agreements?
> > 
> > Most people who work on a GNU software project have a marginal understanding
> > of the legalities, and it is reasonable to believe that there will at least
> > be some percentage of contributors to whom this comes as a complete shock.
> 
> Would seem odd if they signed the disclaimer that there should be any
> surprise.

Odd, our IP lawyer has always suggested that being as explicit as possible
is always a good idea, precisely to avoid any misunderstandings.

> > Further, that really does seem to fly in the face of the spirit of the GPL,
> > and this is touched on by the GPL FAQ:
> > 
> > http://www.gnu.org/licenses/gpl-faq.html#TOCReleaseUnderGPLAndNF
> > 
> > http://www.gnu.org/licenses/gpl-faq.html#TOCDeveloperViolate
> > 
> > That'd be the FSF calling this both "ethically tainted" and showing a loss
> > of "moral standing".  I'd be happy to put it to them to see if there is a
> > more specific opinion covering the case where a copyright holder actually 
> > forces contributors to sign away their rights.
> 
> No one is being forced to sign it over. It is only a requirement for
> those who want their patches merged with the main Digium maintained
> tree.

You contradict yourself.  If you want to be a contributor to Asterisk, you
must sign the agreement.  You cannot be a contributor to Asterisk without
signing the agreement.

> There is nothing stoping anyone from maintaining their own patch
> set seperate of the main tree. 

That's not contributing to the project.

> To an extent, screw the FSF's opinion on this.

Really?  Be very careful.  Once you say that, you really begin to slide 
back down from Mount Principles into the depths of license evil.

> They aren't trying to make this project work nor pay the bills of
> the company that is. 

Well, I've been saying that Stallman's on crack for years, as have many
others, but GPL advocates always seem to be wearing the rose colored
glasses where all resources are free and there's never a reason to charge.
Surely, Stallman in his tenured MIT position has very little clue what 
it's like out in the real world, where one has to pay for stuff...

But if you're going to adopt the GPL, and then you're going to cut a big
hole in it, then I don't think it's wrong to at least discuss it, in a
variety of contexts, including that which would likely be promoted by the
FSF.

> In the same line where they say it is "ethically
> tainted", they also say the copyrightholder can do what ever they want.

Of course, because that's legal fact.

> > > > For all of the people who wanted to tell us about how horrible the BSD
> > > > license is, please explain how this state of affairs is any better.
> > > 
> > > (My memory is spotty and I am not invoking the thread) 
> > > This is like a benevolent dictator, in as much as the only person
> > > allowed to make a proprietary version is Mark/Digium. That is how it is
> > > better. I choose as an option to allow Digium that special right as a
> > > sode effect of merging and maintaining the patch I needed in asterisk.  
> > 
> > Actually, no, Mark/Digium is not the only one allowed to make a propietary
> > version.  Licensing doesn't work that way.  Mark/Digium can assign a license
> > to do whatever to whoever, for whatever reason (with legal caveats that can
> > not be summed up in a box of paper, much less 82 characters).
> > 
> > > > > publicly admit the fact that they have repackaged Asterisk and 
> > > > > contribute enhancements to Asterisk back to the GPL.
> > > > 
> > > > They are not required to contribute changes back.  They are merely
> > > > required to disclose the source code for the Asterisk portion of their
> > > > product.
> > > 
> > > Incorrect. They must disclose any asterisk modifications as it is the
> > > running asterisk code at least to the customer.
> > 
> > Incorrect again.  Go read the GPL.
> > 
> > http://www.gnu.org/licenses/gpl-faq.html#GPLRequireSourcePostedPublic
> > 
> > They are req

RE: [Asterisk-Users] Motherboard whitelist (was Echo - UK Impedanceproblem with X100P?)

[Rich Adamson]

> > > As far as the motherboard issue, its not just the digium products that
> > > have an issue. If you know of someone that is heavy into using audio
> > > applications (eg, song writers, midi stuff), they have known about the
> > > pci / interrupt latency issues on certain motherboards for a lng
> > > time. Wouldn't doubt those folks maintain a list of what's reasonable
> > > verses unacceptable.
> > >
> > You bring up an excellent point -- Can we use the blacklists and
> > whitelists the audio folk have to help the asterisk community?  They'll
> > have all hte same issues -- shared interrupts, craptastic PCI chipsets,
> > etc...  sounds like a great idea to me. 
> > 
> Why not just get a Sipura SPA-3000 and side-step the motherboard vs. TDM
> card compatibility issue altogether?

The spa-3000 isn't much better in _some_ cases. Check the voxilla 
list for lots of complaints relative to echo. Some of that was recently
addressed, but more needs to be done.



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RE: [Asterisk-Users] BRI in the US

[Brian West]

Check this out www.bkw.org/pri.pdf

That's what SBC charges for PRI here... it's the only option I have right
now.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Bielicki
> Sent: Friday, November 12, 2004 2:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] BRI in the US
> 
> the horribly expensive EICON shit. Although if you just want to
> connect ISDN phones to asterisk you can use european ISDN phones with
> cards from Junghanns.net
> 
> 
> On Fri, 12 Nov 2004 14:10:07 -0600, Brian West <[EMAIL PROTECTED]> wrote:
> > What cards will work with asterisk and BRI in the US?
> >
> > bkw
> >
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> 
> 
> --
> Michael Bielicki
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> Joe Greco wrote:
> > Really?  Wouldn't it be nice, then, if Digium explicitly stated that this
> > was their intention, in their little agreements?
> > 
> 
> [snip]
> 
> > That'd be the FSF calling this both "ethically tainted" and showing a loss
> > of "moral standing".  I'd be happy to put it to them to see if there is a
> > more specific opinion covering the case where a copyright holder actually 
> > forces contributors to sign away their rights.
> 
> Digium doesn't force anyone to do anything.  You are not required to 
> have your patches put into Digium's source tree. 

Which is, to say, the tree which is also the GPL'd Asterisk distribution.

> If you don't like the 
> terms of the license then you don't have to use Asterisk nor do you have 
> to contribute to Asterisk.

Asterisk is put forth as an "open source" telephony solution.

I've just shown that the FSF, the people who wrote the license in question,
seem to have a certain disdain for this particular use.

Your only answer is "so don't use it"?

I'll take that as a concession.

> Which of the two DIFFERENT Digium disclaimers do you have a problem with?
> 
> http://www.digium.com/disclaim.changes
> or
> http://www.digium.com/disclaimer.txt
> 
> Anyone that gets caught by suprize should have read the disclaimer more 
> carefully.

The first one, in particular, isn't very clear on what is happening "under
the sheets."  There are a number of open source projects which make you
sign various disclaimers, just to make sure that the contribution is clear
of any work-made-for-hire claims, etc.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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[Asterisk-Users] Can someone tell me what is going on from this debug?

[Doug Eubanks]

Can someone tell me why Asterisk is sending 404 instead of passing this call to 
the demo?  I have replaced the IPs with descriptions

This is the actual asterisk debug,

Thanks
Doug Eubanks
[EMAIL PROTECTED]

Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP GATEKEEPER:5060;branch=z9hG4bK-506011002837883257484230
Via: SIP/2.0/UDP ANALOGADAPTER:5060;branch=z9hG4bK-9a1dbb20
From: "19993152553" ;tag=9d074f5b40d37cf
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 69
Contact: "19993152553" 
Expires: 240
User-agent: Sipura/SPA2000-1.0.37(e)
Content-Length: 399
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
Record-Route: 

v=0
o=- 292396 292396 IN IP4 GATEKEEPER
s=-
c=IN IP4 GATEKEEPER
t=0 0
m=audio 1690 RTP/AVP 0 8 96 2 97 98 18 101 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 NSE/8000
a=ptime:30
a=sendrecv

15 headers, 18 lines
Using latest request as basis request
Sending to GATEKEEPER : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 100
Peer audio RTP is at port GATEKEEPER:1690
Found description format PCMU
Found description format PCMA
Found description format G726-40
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format G729a
Found description format telephone-event
Found description format NSE
Capabilities: us - 0x60e(GSM|ULAW|ALAW|SPEEX|ILBC), peer - 
audio=0x51c(ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 
0x40c(ULAW|ALAW|ILBC)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'sip.simflex.net'
Looking for 19995551212 in default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP GATEKEEPER:5060;branch=z9hG4bK-506011002837883257484230
Via: SIP/2.0/UDP ANALOGADAPTER:5060;branch=z9hG4bK-9a1dbb20
From: "19993152553" ;tag=9d074f5b40d37cf
To: ;tag=as2a8cca97
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to GATEKEEPER:5060
DomainMailV3*CLI>

Sip read:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP GATEKEEPER:5060;branch=z9hG4bK-506011002837893257484230
Via: SIP/2.0/UDP ANALOGADAPTER:5060;branch=z9hG4bK-9a1dbb20
From: "19993152553" ;tag=9d074f5b40d37cf
To: ;tag=as2a8cca97
Call-ID: [EMAIL PROTECTED]
CSeq: 101 ACK
Max-Forwards: 69
Contact: "19993152553" 
User-agent: Sipura/SPA2000-1.0.37(e)
Content-Length: 0


11 headers, 0 lines
Destroying call '[EMAIL PROTECTED]'



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Re: [Asterisk-Users] SysMaster and GPL Violation

[Joe Greco]

> >Further, that really does seem to fly in the face of the spirit of the GPL,
> >and this is touched on by the GPL FAQ:
> >
> >http://www.gnu.org/licenses/gpl-faq.html#TOCReleaseUnderGPLAndNF
> 
> "To release a non-free program is always ethically tainted"
> 
> What's this guy smoking? Oh well, let's let Stallman judge Digium. Good
> idea. Not. 

Well, you have to remember that the GPL is a product of Stallman.  :-)

> >http://www.gnu.org/licenses/gpl-faq.html#TOCDeveloperViolate
> 
> "The developer itself is not bound by it, so no matter what the developer
> does, this is not a "violation" of the GPL."
> 
> Again, let Stallman call out nasal demons against whoever he wants.
> Meanwhile, I think Digium should be able to do whatever they want with code
> they've made, or code disclaimed to them.
> 
> >>That'd be the FSF calling this both "ethically tainted" and showing a loss
> >>of "moral standing".  I'd be happy to put it to them to see if there is a
> >>more specific opinion covering the case where a copyright holder actually 
> >>forces contributors to sign away their rights.
> 
> Where does Digium FORCE people to sign away rights? No one is FORCED to
> contribute the code to Digium. You're fine to make your self-maintained
> version and do whatever you'd like.

Well, see what I said.  If you want to be a contributor to this allegedly
"free software" project, you are forced to sign away your rights.

I'm struggling to think of another free software project where contributed
code bearing an identical GPL or BSD license would require any such
additional disclaimer.

> >Actually, no, Mark/Digium is not the only one allowed to make a propietary
> >version.  Licensing doesn't work that way.  Mark/Digium can assign a
> >license
> >to do whatever to whoever, for whatever reason (with legal caveats that can
> >not be summed up in a box of paper, much less 82 characters).
> 
> Yes, but only Mark/Digium can assign that right. Nothing surprising.

Yes, but that's not what the original poster said.

> >Now, once we finish correcting your statements, we wind up back at my
> >original statement:
> >
> >They are not required to contribute changes back.  They are merely
> >required to disclose the source code for the Asterisk portion of their
> >product.
> 
> Sure, so they disclose that back to their customers, who can then contribute
> it back, if they so desire. I believe that was the intent of the poster.

The intent of the poster is a nice thing indeed, but really one shouldn't
go saying things drastically different about the requirements of the GPL
from the reality of the GPL in defense of the GPL.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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[Asterisk-Users] ACD queue timeout problem

[Goutam Shaw]

 

Hi 

I’m having a timeout problem from the ACD queue. Here
is what I am trying to achieve.

- 
Assign a timeout for the caller entering the ACD
queue

- 
After the timeout put the caller into the voice
mailbox

- 
Let the caller leave the queue by hitting *

 

Here are the settings. The 60 seconds timeout in the
Queue cmd doesn’t seem to work and the caller times out after ringing 2
extensions where as we have 4 active member agents. Also hitting * didn’t get
rid of the call (according to H option in the Queue cmd). Any suggestions to what
I am doing wrong or how it should be done right would be much appreciated.

 

;extensions.conf

;

; HELP-DESK ACD

[macro-sipext-helpdesk]

exten => s,1,Queue(help-desk-queue|Htn|||60)

; Try to send it to a Voicemail upon agents’ unavailability

exten => s,2,Voicemail(u222)

 

 

queues.conf

-

; Our help-desk-queue

[help-desk-queue]

leavewhenempty = yes

music = default

strategy = rrmemory

timeout = 6

retry = 5

maxlen = 0

 

member => SIP/291

member => SIP/292

member => SIP/293

member => SIP/296

 

 

 

Regards,

Goutam Shaw

 

 

 

 

 






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[Asterisk-Users] Answer Confirmation "c"

[Me]

On this page in the Wiki:
http://www.voip-info.org/wiki-Asterisk+ZAP+Channels
This text exist:
*
If the letter c follows, then "Answer Confirmation" is requested, in which 
the call is not considered answered until the called user presses #.
*

Question:
From what I understand you can only use the 'c' option on Zap channels. Is 
there something similar that would allow answer confirmation on NON Zap 
channels?

Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com 

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Re: [Asterisk-Users] DISA() context restrictions

[Michael George]

On Thu, Nov 11, 2004 at 10:58:37AM -0600, Michael Greb wrote:
> On Thu, 11 Nov 2004 09:33:29 +0200 (SAST), [EMAIL PROTECTED]
> <[EMAIL PROTECTED]> wrote:
> > On Tue, 9 Nov 2004, Michael George wrote:
> > 
> > > The only difference to my extensions.conf file is that if I have:
> > > exten => s,2,DISA(no-password, disa)
> > >
> > >
> > > -- Executing DISA("IAX2/[EMAIL PROTECTED]/6", "no-password| disa") in 
> > > new
> > > stack
> > > Nov  9 19:50:33 DEBUG[14521]: app_disa.c:160 disa_exec: Context:  disa
> > 
> > Bet you its the space after the comma.  Notice that the "Context:  disa"
> > has two spaces.
> > 
> > So try DISA(no-password,disa) without the space and see if that helps.
> > 
> > If it does, its obviously a bug, but you have a work-around at least.
> > 
> > Steve
> 
> I wouldn't really call that a bug, especially since I've seen cautions
> in several places against including spaces.  It's just the way it is,
> one wouldn't include spaces in a CSV file, nor inbetween comma
> seperated values in the GECOS field in /etc/passwd, so why between
> arguments in the dial plan.  No fault of Michael George of course, he
> didn't know that was the case before but now he does... I just
> wouldn't call it a bug.

I agree, not necessarily a bug.  It would be nice if the spaces could be
there, but that's just how it is.  Now I know and hopefully others will pick
it up even more easily in teh archives.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[Andrew Kohlsmith]

On November 12, 2004 03:24 pm, Tony Nichols wrote:
> Why not webmin? someone started the interface int (it's in the downloads
> folder; many admins use it, there are s many plugins for it
> currently --- so it must not be THAT hard to code.

Because webmin is unbelievably ugly?

-A.
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Re: [Asterisk-Users] Quick call group question...

[Steve Totaro]

On Fri, 2004-12-11 at 12:01 -0800, Sean Kennedy wrote:
I remember the syntax is something like this: dial(
SIP/1&SIP/2&SIP/3...yada yada yada ), but what I want to know is this:
If someone answers the extension, does it belong to them at that point?
The other lines will stop ringing, and the other people couldn't pick up
the line?
That is correct.  If calling multiple SIP extensions with Dial(), the
first extension to answer the call gets it.  Other extensions will stop
ringing.
Ryan
This also applies to other channels such as IAX or Zap, if used.
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RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?

[Martin List-Petersen]

Citat Rich Adamson <[EMAIL PROTECTED]>:

> No offense intended, but the x100p card was never marketed in the UK (or
> other countries) by digium, etc. Those that found a way to purchase the
> card obviously didn't do the research necessary to understand what they
> were getting into and probably didn't have any sort of understanding of
> what their country telco standards actually were from a system engineering
> perspective. Those that elected the clones obviously have even less
> knowledge/experience.

Sorry, but this need to be correct. Digium does sell and marked the X100P in the
UK. TelAppliant is a official Digium Distributor and is selling the X100P in the
UK. So how more official does it need to be ?

They however don't sell the FXO modules for the TDM cards yet.

Of course, nobody at Digium or TelAppliant could be held responsible for trouble
with a clone card, but that the card never was markedet in the UK is wrong. It
still is !!

Kind regards,
Martin List-Petersen
-- 
"Necessity is the mother of invention" is a silly proverb.  "Necessity
is the mother of futile dodges" is much nearer the truth.
-- Alfred North Whitehead

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Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[Tony Nichols]

On Fri, 2004-11-12 at 14:48, JAMES BOTHAM wrote:
> Hi there,
> 
> I agree with Greg and also with the documentation
> group, we are all great at bitching about * (I know I
> have done a lot of it but thats because UK and support
> for us is minimal or so it feels) we need to unite,
> the only reason Microsoft are so popular is because it
> take 2 minutes to install and applications are usually
> it is easy to configure (coming from Windows to Suse
> was quite easy due to YAST but then from Suse to
> Fedora Core is a nightmare thank god for web min)
> users a nd administrators don't want to be editing
> conf files to do the smallest thing i.e. create a
> dialplan  thats a nightmare, coming from an Avaya
> background (although it has been a year since i
> touched an Avaya INDeX) you could create powerful and
> effective dial plans completely graphically it was so
> easy anybody could do it. Although all system
> administration was done through a menu driven console
> and that was really simple to use. We need to take the
> good from other systems and merge this to Asterisk.
> Also we have to document it. theres no point in
> writing the code if nobody can use it.
> 
> I would like to offer my skills to the production of
> this I can document, bug test and am great at user
> interface design I come from a software house
> background which I can also utilise to get this
> project off the ground.
> 
> I suggest that we all meet in a chat room to create
> some form of a project map and get this off the
> ground.
> 
> 
> Cheers
> 
> James

Why not webmin? someone started the interface int (it's in the downloads
folder; many admins use it, there are s many plugins for it
currently --- so it must not be THAT hard to code.

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Re: [Asterisk-Users] BRI in the US

[Michael Bielicki]

the horribly expensive EICON shit. Although if you just want to
connect ISDN phones to asterisk you can use european ISDN phones with
cards from Junghanns.net


On Fri, 12 Nov 2004 14:10:07 -0600, Brian West <[EMAIL PROTECTED]> wrote:
> What cards will work with asterisk and BRI in the US?
> 
> bkw
> 
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-- 
Michael Bielicki
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Re: [Asterisk-Users] Quick call group question...

[Ryan Courtnage]

On Fri, 2004-12-11 at 12:01 -0800, Sean Kennedy wrote:

> I remember the syntax is something like this: dial( 
> SIP/1&SIP/2&SIP/3...yada yada yada ), but what I want to know is this: 
> If someone answers the extension, does it belong to them at that point?  
> The other lines will stop ringing, and the other people couldn't pick up 
> the line?

That is correct.  If calling multiple SIP extensions with Dial(), the
first extension to answer the call gets it.  Other extensions will stop
ringing.

Ryan


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[Asterisk-Users] BRI in the US

[Brian West]

What cards will work with asterisk and BRI in the US?

bkw

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[Asterisk-Users] Quick call group question...

[Sean Kennedy]

I saw it once in the wiki, but I don't remember what the feature was 
called, so I don't know what to search for.  I know on our current Avaya 
phones, it's called a Call Group.  Where several extensions are dialed 
from one extension ( when an outside line rings, for example ), and the 
person that picks it up gets the call.  Not a Hunt Group, I don't want 
the call to go through a chain of extensions.

I remember the syntax is something like this: dial( 
SIP/1&SIP/2&SIP/3...yada yada yada ), but what I want to know is this: 
If someone answers the extension, does it belong to them at that point?  
The other lines will stop ringing, and the other people couldn't pick up 
the line?

Thanks, sorry if this sounds a little confused, but I am, myself, a 
little confused.  So it all works out.  ;)

Sean
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Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[James Taylor]

Great, count me in.
I'll help with specs, docs, whatever.
I use a Linux based router  for WiFi & T1's: Mikrotik
Burn a CD, boot, install, load the "winbox" on a windows machine and start  
configuring.

Ever try to config a MAX TNT via telnet?
There's now a Maxxmaster GUI (windows).
I can only hope that the two Asterisk boot & install CD's that are out  
there, boost hardware sales for Digium.

I would anticipate that a really good "management" application would help  
sales as well.

I don't think that any Asterisk consultants will suffer from a really good  
management app.  You still have to know what you are doing.

James Taylor
MetroTel
On Fri, 12 Nov 2004 19:48:02 + (GMT), JAMES BOTHAM  
<[EMAIL PROTECTED]> wrote:

Hi there,
I agree with Greg and also with the documentation
group, we are all great at bitching about * (I know I
have done a lot of it but thats because UK and support
for us is minimal or so it feels) we need to unite,
the only reason Microsoft are so popular is because it
take 2 minutes to install and applications are usually
it is easy to configure (coming from Windows to Suse
was quite easy due to YAST but then from Suse to
Fedora Core is a nightmare thank god for web min)
users a nd administrators don't want to be editing
conf files to do the smallest thing i.e. create a
dialplan  thats a nightmare, coming from an Avaya
background (although it has been a year since i
touched an Avaya INDeX) you could create powerful and
effective dial plans completely graphically it was so
easy anybody could do it. Although all system
administration was done through a menu driven console
and that was really simple to use. We need to take the
good from other systems and merge this to Asterisk.
Also we have to document it. theres no point in
writing the code if nobody can use it.
I would like to offer my skills to the production of
this I can document, bug test and am great at user
interface design I come from a software house
background which I can also utilise to get this
project off the ground.
I suggest that we all meet in a chat room to create
some form of a project map and get this off the
ground.
Cheers
James
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[Asterisk-Users] Asterisk crashes after call when running as non-root, bug???

[Joost Kraaijeveld]

Hi all,

I am using Debian Sarge (2.6) with ISDN4Linux. If I run asterisk as root 
everyting is OK. If I run asterisk as the user asterisk, the programm crashes 
after answering a call. No message in the asterisk logs but the 
/var/log/messages says:

Nov 12 20:42:01 localhost kernel: isdn: HiSax1,ch0 cause: E0010

Is this a bug or is this a known feature that can be solved by properly 
configuring asterisk?

TIA

Joost
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Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[Gregory Junker]

I've registered channel #acd on Freenode if anyone wants to pop in.
Greg
JAMES BOTHAM wrote:
Hi there,
I agree with Greg and also with the documentation
group, we are all great at bitching about * (I know I
have done a lot of it but thats because UK and support
for us is minimal or so it feels) we need to unite,
the only reason Microsoft are so popular is because it
take 2 minutes to install and applications are usually
it is easy to configure (coming from Windows to Suse
was quite easy due to YAST but then from Suse to
Fedora Core is a nightmare thank god for web min)
users a nd administrators don't want to be editing
conf files to do the smallest thing i.e. create a
dialplan  thats a nightmare, coming from an Avaya
background (although it has been a year since i
touched an Avaya INDeX) you could create powerful and
effective dial plans completely graphically it was so
easy anybody could do it. Although all system
administration was done through a menu driven console
and that was really simple to use. We need to take the
good from other systems and merge this to Asterisk.
Also we have to document it. theres no point in
writing the code if nobody can use it.
I would like to offer my skills to the production of
this I can document, bug test and am great at user
interface design I come from a software house
background which I can also utilise to get this
project off the ground.
I suggest that we all meet in a chat room to create
some form of a project map and get this off the
ground.
Cheers
James
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Re: [Asterisk-Users] No ringing with Phonejack Lite - hardware or software problem?

[Michael Vogel]

Michael Vogel schrieb:
Does everything looks okay in your eyes? If it seems okay I will look if 
my telephone has a failure (maybe the ring voltage is too low?) or the 
card is defective?
The card isn't defective, the software is allright. It was my phone that 
couldn't handle with the ring voltage. Now I've got a new phone that 
rings when I phone it.

Bye!
Michael
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Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[JAMES BOTHAM]

Hi there,

I agree with Greg and also with the documentation
group, we are all great at bitching about * (I know I
have done a lot of it but thats because UK and support
for us is minimal or so it feels) we need to unite,
the only reason Microsoft are so popular is because it
take 2 minutes to install and applications are usually
it is easy to configure (coming from Windows to Suse
was quite easy due to YAST but then from Suse to
Fedora Core is a nightmare thank god for web min)
users a nd administrators don't want to be editing
conf files to do the smallest thing i.e. create a
dialplan  thats a nightmare, coming from an Avaya
background (although it has been a year since i
touched an Avaya INDeX) you could create powerful and
effective dial plans completely graphically it was so
easy anybody could do it. Although all system
administration was done through a menu driven console
and that was really simple to use. We need to take the
good from other systems and merge this to Asterisk.
Also we have to document it. theres no point in
writing the code if nobody can use it.

I would like to offer my skills to the production of
this I can document, bug test and am great at user
interface design I come from a software house
background which I can also utilise to get this
project off the ground.

I suggest that we all meet in a chat room to create
some form of a project map and get this off the
ground.


Cheers

James

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Re: [Asterisk-Users] Combination Cellular and WiFi/SIP

[James Taylor]

I get RCR and last month, evrybody said "It's coming".
Blackberry...
http://www.engadget.com/entry/2227024145845547/
A friend of mine runs Nextel on his AND he likes it.
So, you take the $50/month unlimited free incoming plan, program Asterisk  
for callback - no answer - no supervision - phone my mobile...

James Taylor
MetroTel
On Fri, 12 Nov 2004 14:21:57 -0500 (EST), Andrew McRory  
<[EMAIL PROTECTED]> wrote:

It's time to replace my aging kyocera phone and I have been looking for
something that supports both cell and sip. Everything I read says it's
coming from major manufacturers soon, etc. So waht are the options? Get a
pda/phone and run xlite or what?
Thanks,

--
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Re: [Asterisk-Users] Motherboard whitelist (was Echo - UK Impedanceproblem with X100P?)

[Andrew Kohlsmith]

On November 12, 2004 02:07 pm, Kevin Walsh wrote:
> Why not just get a Sipura SPA-3000 and side-step the motherboard vs. TDM
> card compatibility issue altogether?

I prefer to have my phone system in a computer where it's all configurable and 
"local"...  I dislike wall-warts and small plastic boxes and most of all I 
despise SIP.

If you just want a box that you can plug in somewhere and go and you're happy 
with the functionality therein, I don't see any troubles with it.  :-)  I 
don't know of a Sipura that terminates 1-4T1s though.  :-)

-A.
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Re: [Asterisk-Users] $200 AMP documentation bounty

[Jason Becker]

dean collins wrote:
There is a $200 bounty for helping document a step by step guide to AMP, 
anyone on this list interested in making easy money feel free to contact me.
I want to thank Dean for giving me this opportunity to address some of 
the concerns raised in regards to AMP.

AMP began as a value-add to a _small business_ turn-key solution. We 
felt strongly that in order to approach small businesses with our 
integrated solution we needed a GUI administration interface that 
handled the mundane. When we looked at what was available circa 
Feb/March of this year we realized that we would be better off writing 
it ourselves. Most of the other (freely available) candidates still 
appear to me to be thin wrappers to the configuration files that require 
the end user to have a knowledge of Asterisk and programming logic. From 
the outset, AMP - or more specifically our piece that allows for setup 
of extensions, IVRs, etc. - was designed so that an Office Manager of a 
small business could use it for the day-to-day operational stuff.

It's clear to me that our design goal has resulted in the attention of 
people whose expertise does not extend to the Linux/UNIX world. I 
personally am thrilled at this. I also understand the frustration these 
people must feel when met with the INSTALL instructions. We know that if 
a person has software installation / development experience on Linux 
that a fully functioning AMP can be installed in a couple hours. We 
cannot take responsibility for the Linux learning curve. As one poster 
to this thread commented, "the best way to overcome it is to take a 
crash course in general Linux admin". I think this is good advice for 
the time being until we find a solution to the problem of providing the 
user base with well-written documentation.

As to the specifics of providing the said documentation I am open to 
suggestions. Open source economics make it challenging for the software 
provider to also provide the other elements of a mature product. 
Elements like documentation and training. In fact, part of the reason we 
open sourced our piece of AMP was so that we could solicit help from the 
 Asterisk community to service this need. asteriskdocs.org was a 
suggestion. We, Coalescent Systems, would also consider hiring a 
professional technical writer if we can ensure that the demand for 
documentation was significant and that purchase of the documentation 
(ala JBoss' model) would support the paid position.

AMP is still targeted towards the small office and small business. I 
think we have made that very clear on the AMP homepage:

http://amp.voxbox.ca
In short, AMP, by design, limits the feature set of Asterisk and is 
intended for small office and small business use. It does NOT align well 
with the needs of larger organizations and is not suited for use in 
those environments. If Gregory Junker et al. want to design a 
"standalone, cross-platform" alternative I applaud their initiative.

Finally, I want to thank everyone that has shown an interest in AMP and 
Coalescent Systems.

Regards,
--
Jason Becker
Director & CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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RE: [Asterisk-Users] astGUIclient - 1.0.4 (Running in Windows) and SQLUpdater Down

[Guido Rebert]

I´m happy, I finally did it!!!  sweet!!
What about you?

Guido Rebert
Network Manager
GrupoPyD - +54 11 4800


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ken Chan
Enviado el: Viernes, 12 de Noviembre de 2004 10:45 a.m.
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] astGUIclient - 1.0.4 (Running in Windows) and
SQLUpdater Down


Hello,
I just followed the SCRATCH INSTALLATION and installed the astGUIclient -
1.0.4 in my Window PC.

After I doubled click the iron and the window comes up.  After a few
seconds, it complained that "SQL Updater" is down.  (Attached please found
the error messages shown in my Windows PC).

Anyway, checked the Asterisk PC (ps -aux) and both
"/usr/local/mysql/bin/mysqld_safe --user=mysql" and "[mysqpld]" are still
running.

Any idea where my problem is?

Thanks
ken





Updater down!!!

getpeername() on closed socket GEN31 at C:/Perl/lib/IO/Socket.pm line 199.
getpeername() on closed socket GEN31 at C:/Perl/lib/IO/Socket.pm line 218.
send() on closed socket GEN31 at C:/Perl/lib/IO/Socket.pm line 218.
recv() on closed socket GEN31 at C:/Perl/lib/IO/Socket.pm line 236. substr
outside of string at C:\AST_VICI\libs/Net/MySQL.pm line 425. Use of
uninitialized value in unpack at C:\AST_VICI\libs/Net/MySQL.pm line 425.
getpeername() on closed socket GEN31 at C:/Perl/lib/IO/Socket.pm line 199.
 error:send: Cannot determine peer address at C:\AST_VICI\libs/Net/MySQL.pm
line  100

Tk::Error: send: Cannot determine peer address at
C:\AST_VICI\libs/Net/MySQL.pm line 100  Carp::croak at C:/Perl/lib/Carp.pm
line 191  IO::Socket::send at C:/Perl/lib/IO/Socket.pm line 215
Net::MySQL::close at C:\AST_VICI\libs/Net/MySQL.pm line 100
main::get_online_channels at C:\Documents and
Settings\kchan\Desktop\astGUIclie nt_1.0.4.pl line 2105  Tk::After::repeat
at C:/Perl/site/lib/Tk/After.pm line 79
[repeat,[{},after#157,1000,repeat,[\&main::get_online_channels]]]
 ("after" script)






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[Asterisk-Users] Combination Cellular and WiFi/SIP

[Andrew McRory]

It's time to replace my aging kyocera phone and I have been looking for 
something that supports both cell and sip. Everything I read says it's 
coming from major manufacturers soon, etc. So waht are the options? Get a 
pda/phone and run xlite or what?

Thanks,

-- 
Andrew McRory - President/CTO 
Linux Systems Engineers, Inc. - http://www.linuxsys.com
Located in beautiful Tallahassee, Florida
Office  850-224-5737
Office  850-575-7213
Mobile  850-294-7567


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Re: [Asterisk-Users] Re: Top posting

[Scott Lykens]

On Fri, 12 Nov 2004 18:57:05 -, Kevin Walsh <[EMAIL PROTECTED]> wrote:

> Perhaps you should adopt a more flexible attitude and learn to follow
> up properly.  There is no excuse at all for lazily top-posting.

This from the resident signature size champion.

Lay off of it.

I've seen this goofy flame several times from you, every time
accompanied by your childish and egotistical signature, but do you see
us flaming you for it every time you post?

People will do things that annoy you, that's life. Perhaps you should
adopt a more flexible attitude in accepting other people's
idiosyncrasies instead of flaming everyone who may cause you distress
by top posting.
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Re: [Asterisk-Users] SysMaster and GPL Violation -- what about IPEYA

[Andrew Thompson]

Mark Spencer wrote:
There seems to be some confusion here so I would like to make a few 
brief comments and will likely not add much to this thread other than 
these few things:

1) Digium *does* license Asterisk (as we distribute it, no additional 
features) outside of GPL and we *do* have commercial licensees already.

2) Digium appreciates the community keeping a watchful eye on other 
products in the marketplace which may be in violation of Asterisk's 
licensing terms.  Please feel free to contact us directly if you have 
any concerns or questions.

3) I do not wish to comment specifically about Sysmaster's relationship 
with Digium at this time other than to say we are in contact with them.

Thank you again for all of your support in the community.
Mark
There, you have an answer. Digium is in contact with Sysmaster. It will 
be resolved one way or another.

This is where, on a web-based forum, the Admin would lock the thread... 
Please, can we call this thread closed?

--
Andrew Thompson
http://aktzero.com/
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RE: [Asterisk-Users] Re: Top posting

[Kevin Walsh]

Richard Cook [EMAIL PROTECTED] lazily top-posted:
> Must be nice to have time and money to worry about someone's posting
> method. Amazing. 
> 
Time is money, and I will waste neither trying to work out what it is
you just followed up to.  Perhaps if you learned to post in context...

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RE: [Asterisk-Users] SysMaster and GPL Violation -- what about IPEYA

[Mark Spencer]

There seems to be some confusion here so I would like to make a few brief 
comments and will likely not add much to this thread other than these few 
things:

1) Digium *does* license Asterisk (as we distribute it, no additional 
features) outside of GPL and we *do* have commercial licensees already.

2) Digium appreciates the community keeping a watchful eye on other 
products in the marketplace which may be in violation of Asterisk's 
licensing terms.  Please feel free to contact us directly if you have any 
concerns or questions.

3) I do not wish to comment specifically about Sysmaster's relationship 
with Digium at this time other than to say we are in contact with them.

Thank you again for all of your support in the community.
Mark
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RE: [Asterisk-Users] Motherboard whitelist (was Echo - UK Impedanceproblem with X100P?)

[Kevin Walsh]

> On November 12, 2004 09:46 am, Rich Adamson wrote:
> > As far as the motherboard issue, its not just the digium products that
> > have an issue. If you know of someone that is heavy into using audio
> > applications (eg, song writers, midi stuff), they have known about the
> > pci / interrupt latency issues on certain motherboards for a lng
> > time. Wouldn't doubt those folks maintain a list of what's reasonable
> > verses unacceptable.
> >
> You bring up an excellent point -- Can we use the blacklists and
> whitelists the audio folk have to help the asterisk community?  They'll
> have all hte same issues -- shared interrupts, craptastic PCI chipsets,
> etc...  sounds like a great idea to me. 
> 
Why not just get a Sipura SPA-3000 and side-step the motherboard vs. TDM
card compatibility issue altogether?

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RE: [Asterisk-Users] SIP & ALERT_INFO for distinctive ring

[Brian West]

http://bugs.digium.com/bug_view_page.php?bug_id=0002838

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kubat, Philip
> Sent: Friday, November 12, 2004 12:08 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] SIP & ALERT_INFO for distinctive ring
> 
> Agreed, works fine on creating calls, e.g outbound.
> 
> What I am looking for; is there means to capture this (maybe via a
> variable)
> on call into Asterisk?
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Friday, November 12, 2004 12:43 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] SIP & ALERT_INFO for distinctive ring
> 
> You need ot set "_ALERT_INFO"  and yes it works.
> 
> bkw
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Kubat, Philip
> > Sent: Friday, November 12, 2004 11:33 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: [Asterisk-Users] SIP & ALERT_INFO for distinctive ring
> >
> > Does anyone have SIP distinctive rings working with SIP providers,
> > inbound?
> > BroadVoice allows for several numbers on a single account, which they
> > delivered with distinctive ring over the primary number.  All the calls
> > come
> > in with the sip header “from” as the primary number.  It looks like (via
> > sip
> > debug and ethereal) that the SIP header variable “ALERT_INFO” is set to
> a
> > ringer type.  (I believe this is part of the RFC)
> >
> > >From what I can figure Asterisk supports setting “ALERT_INFO” for
> sending
> > calls to SIP devices.
> >
> > My question is can I read it for inbound calls?
> > Other ideas?
> >
> > Thanks,
> > Philip
> >
> >
> >
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Re: [Asterisk-Users] SysMaster and GPL Violation

[Steven Critchfield]

On Fri, 2004-11-12 at 12:24 -0600, Joe Greco wrote:
> > On Fri, 2004-11-12 at 09:26 -0600, Joe Greco wrote:
> > > > I too demand sysmaster either pay Digium for a non-gpl license or 
> > > 
> > > Now, here, this gets to the heart of a problem I've hinted at before.
> > > 
> > > Digium is making people sign a draconian agreement that gives up rights
> > > to patches and features that are integrated into Asterisk, by signing
> > > rights over to Digium.
> > > 
> > > I would expect that most contributors do not realize that they are setting
> > > up a scenario where Digium can, in fact, sell non-GPL Asterisk licenses 
> > > to 
> > > third parties and essentially sell their work.
> > 
> > I think we all knew that. In fact, we consider it a good thing as it
> > allows Digium an income to keep paid developers working on the code
> > base.
> 
> Really?  Wouldn't it be nice, then, if Digium explicitly stated that this
> was their intention, in their little agreements?
> 
> Most people who work on a GNU software project have a marginal understanding
> of the legalities, and it is reasonable to believe that there will at least
> be some percentage of contributors to whom this comes as a complete shock.

Would seem odd if they signed the disclaimer that there should be any
surprise.

> Further, that really does seem to fly in the face of the spirit of the GPL,
> and this is touched on by the GPL FAQ:
> 
> http://www.gnu.org/licenses/gpl-faq.html#TOCReleaseUnderGPLAndNF
> 
> http://www.gnu.org/licenses/gpl-faq.html#TOCDeveloperViolate
> 
> That'd be the FSF calling this both "ethically tainted" and showing a loss
> of "moral standing".  I'd be happy to put it to them to see if there is a
> more specific opinion covering the case where a copyright holder actually 
> forces contributors to sign away their rights.

No one is being forced to sign it over. It is only a requirement for
those who want their patches merged with the main Digium maintained
tree. There is nothing stoping anyone from maintaining their own patch
set seperate of the main tree. To an extent, screw the FSF's opinion on
this. They aren't trying to make this project work nor pay the bills of
the company that is. In the same line where they say it is "ethically
tainted", they also say the copyrightholder can do what ever they want.

> > > For all of the people who wanted to tell us about how horrible the BSD
> > > license is, please explain how this state of affairs is any better.
> > 
> > (My memory is spotty and I am not invoking the thread) 
> > This is like a benevolent dictator, in as much as the only person
> > allowed to make a proprietary version is Mark/Digium. That is how it is
> > better. I choose as an option to allow Digium that special right as a
> > sode effect of merging and maintaining the patch I needed in asterisk.  
> 
> Actually, no, Mark/Digium is not the only one allowed to make a propietary
> version.  Licensing doesn't work that way.  Mark/Digium can assign a license
> to do whatever to whoever, for whatever reason (with legal caveats that can
> not be summed up in a box of paper, much less 82 characters).
> 
> > > > publicly admit the fact that they have repackaged Asterisk and 
> > > > contribute enhancements to Asterisk back to the GPL.
> > > 
> > > They are not required to contribute changes back.  They are merely
> > > required to disclose the source code for the Asterisk portion of their
> > > product.
> > 
> > Incorrect. They must disclose any asterisk modifications as it is the
> > running asterisk code at least to the customer.
> 
> Incorrect again.  Go read the GPL.
> 
> http://www.gnu.org/licenses/gpl-faq.html#GPLRequireSourcePostedPublic
> 
> They are required to /make it available/, but they are not under some sort
> of obligation to proactively provide it to customers (sec 3 sub B).

So the only part that isn't explicilty correct in my comment above is
the assumption that some customer would seek it out. You have assumed
the other option.

> > All modifications of the
> > code are forced to be covered by the GPL. The customer at this point has
> > the opertunity to then contribute those changes back to the community as
> > the GPL explicitly allows redistribution. The difference being that the
> > company in question doesn't have to distribute the changes back to us,
> > but they have to distribute them to their clients.
> 
> No, they merely need to make them available.  You could correct your
> statement by saying "..., but they have to be willing to distribute them."
> 
> Note specifically that I have struck "to their clients", because that is 
> not the way the GPL works (sec 3 sub B).  Because the GPL grants the 
> holder the right to further distribute it, the responsibility to be 
> willing to distribute is not limited in the way you suggest.
> 
> Now, once we finish correcting your statements, we wind up back at my
> original statement:
> 
> They are not required to contribute changes back.  They are merely

RE: [Asterisk-Users] Re: Top posting

[Brian West]

It must be nice to waste everyone's time and bandwidth with such idle
bullshit.

bkw



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