k.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://li
80
5 8088
1 8443
1 873
1 8889
1 9124
2 9191
I hope those of you with internet accessible systems are following best
practices!
murf
--
Steve Murphy
--
_
-- Bandwidth and Colocation Provid
SOLVED!
Many THANKS to George and Anthony! See at the very end, my comments...
On Thu, Sep 8, 2016 at 5:58 PM, Anthony Joseph Messina <
amess...@messinet.com> wrote:
> On Thursday, September 8, 2016 1:12:36 PM CDT Steve Murphy wrote:
> > Hello!
> >
> > Oh, wise on
(G.729 replaced with alaw)
direct_media=no
context=phone
rtp_timeout=120
set_var=__phoneid=12
set_var=__contacttypeid=4
set_var=__phonelineid=78
callerid="Steve Murphy" <101>
call_group=2
pickup_group=2
mailboxes=101@murftest
language=en
send_rpid=yes
send_pai=yes
OK, that complet
decs will not load on Asterisk 13.
a "few weeks" has turned into almost half a year now. Are these
codecs no longer going to be available for 13 and up? Or, were they
just overlooked in the day-to-day rush called life?
murf
--
Steve Murphy
ParseTree Corporation
--
vision.
5. Keep your logs for a couple years.
6. Change your phone SIP acct passwords now, if you haven't
implemented the above precautions yet.
If I missed a previous post on this, forgive me.
Just thought you-all might appreciate a heads-up.
murf
--
Steve Murphy
ParseTree Corporat
rs will not
find you. Seems a much simpler solution... but there are some drawbacks...
can anyone think of them? And will these drawbacks matter to you? And, given
this solution, will the odds that a scanner might find your machine be so
low,
that it is not worth using something like fail2ban to
and some scripts to
install them.
I'd like to help with at least one new language, if possible, to see
what can be done to smooth the process.
Interested? Write me!
murf
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dot com
☎ 307-899-5535
--
__
On Wed, Dec 11, 2013 at 3:29 PM, Matthew Jordan wrote:
>
> On Wed, Dec 11, 2013 at 3:15 PM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>
>> On 13-12-11 03:15 PM, Steve Murphy wrote:
>>
>>> I see the following paragraph in the Asterisk t
languages easily findable and downloadable?
I see a good selection on
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
murf
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf att parsetree dott com
☎ 307-899-5535
that Asterisk distributes. I would assume that
the source sounds are all 44khz (or more) cd quality sounds, probably in
pcm wave format, maybe even in stereo. Are these source
sound sets indeed withheld from the public? Or am I mistaken in my
impression?
murf
--
Steve Murphy
ParseTree Corpor
otocols to
this
list: ARI, and the ExternalIVR interface.
If not, it might be instructive to learn why!
murf
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dott com
☎ 307-899-5535
--
__
fig,
there may be something there that might prevent DTMF. Same with the phone
settings.
Best of Luck,
murf
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dott com
☎ 307-899-5535
--
_
-
th build instructions, via my
project at github:
https://github.com/WyoMurf/SaySentence.git
I am looking for help from developers and translators.
If you are interested in knowing more, have objections, ideas, etc, let me
know!
murf
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82
you want, then call the Voicemail() app for that user. If you
use a few milliseconds of silence in all the recorded message files, then
you have effectively overridden its built-in sound file playing with your
own introductory messages instead.
murf
--
Steve Murphy
--
___
o I need fancy options with the "Dial" command doing
> GoSub and what not? and Why does it insist on playing all these prompts I
> have commented them all out from followme.conf, but it's still looking to
> play them
>
> Thanks in advance
> \A
>
>
> --
> __
se seen this sort of thing? Any words of wisdom?
murf
--
Steve Murphy
ParseTree Corporation
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
because extension not found in
>>> context 'default'.
>>> == Using SIP RTP CoS mark 5
>>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
>>> Call from '' to extension 'service' rejected because extension not fou
list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ m...@parsetree.com
☎ 307-899-5535
--
_
--
tricky to debug. Then we might
resort to
stuff like dmalloc, and others, to help spot where/when corruption occurs.
Let's cross that
bridge if we come to it.
murf
On Tue, Apr 5, 2011 at 7:30 AM, Jerry Geis wrote:
> Jerry Geis wrote:
>
>>
>> Steve Murphy wrote:
>
ion
> (smvoice-mediaport-public-address, s, 3) exited non-zero on
> 'SIP/mndemo_to_mediaport105-0003'
> --
>
>
> As I mentioned starting asterisk all this works. There is some random time
> later - perhaps days where it then stops
> finding the exten.
>
t.
If you know the number of digits, and it is fixed, then you could use
_011870XXX or similar to avoid the timeout, and begin the Goto
immediately on reception of the final digit.
The X in the second set will match just on
show how I tied sipp and asterisk together. It might not at all help
you, might not be your approach at all, but it might give you some ideas.
Best of luck!
murf
--
Steve Murphy
--
_
-- Bandwidth and Colocation Provided by htt
On Sat, Dec 25, 2010 at 7:41 PM, dave george wrote:
> Yes we have that set in logger.conf.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Ustinov
> Sent: Saturday, December 25, 2010 6:25 PM
> To: A
lsof via your package manager)
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every T
On Thu, Dec 9, 2010 at 10:31 AM, Daniel Tryba wrote:
>
> You could use SIPVicious to run attacks on your own servers:
> http://code.google.com/p/sipvicious/
>
Yeah, why not? All the criminals on the internet are using it, too! ;^)
I'm seeing 1-4 scans per day on the average. And it's pretty cl
On Wed, Dec 1, 2010 at 12:15 PM, Kevin P. Fleming wrote:
> On 12/01/2010 01:05 PM, Steve Murphy wrote:
> > Hello,
> >
> > I wonder if anyone else has noticed this.
> >
> > I see a pair of calls to pipe() within the codec_g729a, and suddenly, I
> > have a le
Hello,
I wonder if anyone else has noticed this.
I see a pair of calls to pipe() within the codec_g729a, and suddenly, I have
a leaked file descriptor that remains until asterisk dies.
Now, maybe no-one sees this, mainly because I have no g729 licenses on the
machines where this happens. And con
ith
CDR's. If you don't mind losing a little chunk of the conversation here or
there,
the current CDR's should be sufficient for you, and you don't have to go
thru
any bother.
Just keep in mind that clever peo
ff can also be done via the "simple" method, and
so I'm
about half way thru the spec, expunging the "complex" stuff. All my examples
have to be
changed -- If you are interested in looking at my spec, you can:
svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs
and look
or a voip trunk? Or is it just like a serial number for the scan? What?
Here's some examples:
2648061411
3190339404
2685608247
3358171034
2092652562
2206598858
Just trying to follow the advice: "Know thy Enemy"
murf
Steve Murphy
ParseTree Corp.
57 Lane 17
Cody, WY 82414
✉
f, because
it would be in the "addons" stuff
for Asterisk.
But, if you look at the similarities between the CEL backends, and the CDR
backends, you'll probably
notice that you could pump out a myseql backend with the same mods in a
short amount of time.
I would be curious
p://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> ___
A few corrections!
On Tue, Sep 21, 2010 at 6:32 PM, Steve Murphy wrote:
>
>
> On Tue, Sep 7, 2010 at 5:36 PM, Fabiano Carlos Heringer <
> b...@grupoheringer.com.br> wrote:
>
>> Em 07/09/2010 17:15, Miguel Molina escreveu:
>>
>> El 07/09/10 14:49, Fab
svn.digium.com/svn/team/murf/asterisk-RFCs and look at the
document in there (I have a few different formats, the .docx is the source).
It's been in flux. Just the first few examples are accurate. Let me know
what you think.
murf
--
Steve Murphy
ParseTree Corp
--
But, if you only have one or two DID's, all the machinery and programming
seem
a bit overkill.
murf
>
> --
>
--
Steve Murphy
ParseTree Corp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
; -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
> [pbx_config]
>
> [ Included context 'dialout3' created by 'pbx_config' ]
> '_1NXXNXX' => -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
> [pbx_config]
>
> -= 7 extensions (7 priorities) in 7 contexts
ntering:
> >
> > *CLI> dialplan show 6789542...@remote
> >
> >
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital
t; New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Steve Murphy
ParseTree C
_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.o
nation is "s", not a phone
> number?
>
> Thanks for any assistance!
> J
>
> --
> _
> -- Bandwidth and Colocation Pro
asterisk yield
> incorrect alaw files?
>
> Please help, thanks
>
> Quyps
>
> On Thu, 2010-05-20 at 00:50 -0600, Steve Murphy wrote:
> > Quyps--
> >
> > I've noticed in general that the ulaw, alaw, gsm, slin files used and
> > generated by
> > ast
p
> here?
>
> Thanks in advance.
> Quyps
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>
-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _________
__
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or updat
ks?
>
> This was nothing more then PR hype...
>
> Stu
>
>
Assuming that every such spamming/hacking/attack site is funded on a
stolen identity/CC number, it will soon sink into Amazon that they are
getting a bad rep, and losing money on such problems, as all such charges
are reversed
http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-us
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users m
[pbx_config]
> 2. Hangup()
> [pbx_config]
>
>
> As you can see the only "." has been erased.
> There is no problem on DID ("." notations works fine), but only in CID
> field.
> I'm usign Asterisk 1.4.26.2
>
>
>
>
>
> --
> _________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
&
On Tue, Feb 16, 2010 at 3:01 AM, Olle E. Johansson wrote:
>
> 16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:
>
> > On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
> >> On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri
> wrote:
> >>
> >>&
On Tue, Feb 16, 2010 at 1:43 AM, Tzafrir Cohen wrote:
> On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
> > On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri
> wrote:
> >
> > > Yes but in any case you can enter all of the strings that reasonably
>
--
> Loway - home of QueueMetrics - http://queuemetrics.com
>
>
> --
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update
lting, LLC
> Full-stack web design and development
> http://quinnweaver.com/
> 510-520-5217
>
> --
> _____
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mail
gt; http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Steve Murphy
ParseTree Corp
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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1))
>
>
> -Dave
>
> --
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium
ial (imho) impression is that localizations in the code, are in
general a bad way to approach
localization in general. The localizations should be located neither in
Asterisk code nor in dialplan code.
I know, I know, that such code already exists, but that's still not making
my assertion f
_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update option
> Tilghman & Teryl
> with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
> and Harry, BB, & George (dogs)
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mai
>> *Sent:* Friday, May 29, 2009 10:29 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] Attended transfer and dialplan
>>
>>
>>
>> Hi,
>>
>> How can you add specific statements into Asterisk dialplan (e
f
undocumented
behavior, that may be tricky to imitate.
murf
>
>
> Thanks
> Jim
>
>
--
Steve Murphy
ParseTree Corp
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asterisk-users mailing list
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do ?
>
> Regards
>
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/ast
hin Asterisk. Thus, only
external referencing is possible to individual CDR records. But this should
be good
enough, I think!
This may or may not help with issues brought forward a while back
by greyman and others, -- I'd appreciate hearing about it.
murf
--
Steve Murphy, Pres, Consultant
it will play MOH up to the time someone answers. No
ringing/moh mixture...
Dial doesn't do that. You may have to correct some typos, etc. that I've
made above!
A hangup from the remote end will end the Dial app, and the result should be
ANSWE
channel in the channel role. The parking manager doesn't
run the h-exten if a channel hangs up while parked. And channel
and peer roles can sometimes get a bit confused in transfer
scenarios. The truth of each sentence above will surely change
with new releases...
murf
--
Steve Mur
ou want to track; those
involved
with dialing would be automatically tracked. Or time groups of invocations
via
forcing a leg-split via a simple dialplan application call...
again, read the doc, and let me know what you think.
murf
--
Steve Murphy
ParseTree Corp
_
es :
>
> context mylocal {
> includes {
> subs;
> };
>
>700 => ...
> };
>
> Regards
>
> _______
> -- Bandwidth and Colocation Provided by http://www.api-digital.c
On Sat, 2009-02-07 at 15:51 -0500, Alexander Lopez wrote:
>
> > -Original Message-
> > From: Steve Murphy [mailto:m...@digium.com]
> > Sent: Saturday, February 07, 2009 1:59 PM
> > To: Alexander Lopez
> > Subject: RE: [asterisk-dev] 1.4 and CDRs -- The Bre
or
> such calls ?
>
> Rgds
> Sriram
>
Sriram--
Well, if the end or ANSWER time isn't set, then you would get a 0
duration.
murf
--
Steve Murphy
Digium
smime.p7s
Description: S/MIME cryptographic signature
___
-- Bandwidth and Co
),
then I'd get rid of the Hangup() call, because it's useless. (The
h-exten
is being run because of a hangup situation in the first place.)
murf
--
Steve Murphy
Digium
smime.p7s
Description: S/MIME cryptographic signature
___
-- Bandwid
On Tue, 2009-01-13 at 21:09 +0100, Benny Amorsen wrote:
> I wrote a really long email, but it hinged on one thing I need
> clarified...
>
> tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy:
>
> > CDR1: A -> B start: e1a ans: e2 end: e4 Party: B disp:
>
orsen wrote:
> Steve Murphy writes:
>
> > Which of the two would you see being useful to you?
>
> "Leg based", as far as I can see, because that looks like the only way
> to bill transfers differently depending on which end did the transfer.
>
> Possibly "
On Mon, 2009-01-12 at 17:08 -0200, David fire wrote:
>
>
> 2009/1/12 Russell Brown
> Quoth Steve Murphy...
> >Date: Mon, 12 Jan 2009 08:51:03 -0700
> >
> >QUESTIONS:
> >
> >Which of
On Mon, 2009-01-12 at 19:26 +0200, Apostolos Pantsiopoulos wrote:
> Steve Murphy wrote:
> > Hello!
...
> Hi,
>
> The specs look very promising. I think everyone
> here should be grateful for your efforts. In answer to your
> question I personally find both approach
stem would be
deprecated, and in some "futurer" release the "old" (now current)
CDR system would be dropped entirely. What do you
think? Are we high on drugs, or what?
murf
--
Steve Murphy
Digium
smime.p7s
Description: S/MIME cryptographic signature
On Fri, 2009-01-09 at 04:24 +, Grey Man wrote:
> On Fri, Jan 9, 2009 at 3:48 AM, Steve Murphy wrote:
> >
> > But, since it is timestamp based, and unique in that the final part was
> > incremented per request in the same sec, it made a great item to sort
> > on,
e h-exten is,
it's the hangup extension in the dialplan; the pbx will execute
that extension when the channel is being hung up.)... but I won't.
Just my thoughts on the fine points of the to-be-developed
new CDR system...
murf
--
Steve Murphy
Digium
smime.p7s
Description: S/
you immediately
what sys it came from, if you did that.
But it's quite legitimate to want to use UUID's. I have no idea how much
processing power they take to be generated, probably not much. There's
pros and cons...
just a thought,
murf
> Regards,
>
> Greyman.
--
St
nd being a token, and you avoid the syntax error.
You have to keep in mind that by the time the $[ ... ] exprs are evaluated,
all ${..} constructs have been recursively evaluated away.
the wiki is a good reference for AEL2...
(http://voip-info.org/wiki/view/Asterisk+AEL2
and there is a
On Thu, 2009-01-08 at 16:20 +, Grey Man wrote:
> On Wed, Jan 7, 2009 at 6:01 PM, Steve Murphy wrote:
> > On Wed, 2009-01-07 at 02:56 +, Grey Man wrote:
> >> On Tue, Jan 6, 2009 at 3:53 PM, Steve Murphy wrote:
> >>
> >> That sounds a bit dangerous to me
in the CDR fields, now, really, as the conversation
continues...
On Wed, 2009-01-07 at 02:56 +, Grey Man wrote:
> On Tue, Jan 6, 2009 at 3:53 PM, Steve Murphy wrote:
> > As to Answers, we have to start getting pedantic; if A is an exten,
> > then the first answer will be when B
Tue, 2009-01-06 at 10:37 +, Grey Man wrote:
> On Mon, Jan 5, 2009 at 6:42 PM, Steve Murphy wrote:
> > I **think** I have a handle on it... Basically, for each channel
> > that did anything, no matter what, you'd like a single CDR
> > for that channel that would reco
, but
> it's set to 'yes.'
>
> Thanks in advance for the help.
Robert--
Could this be the same as Mantis bug 13691?
(http://bugs.digium.com/view.php?id=13691)
I'm hoping to get some time and try to clear out a bunch of CDR bugs...
murf
--
Steve Murp
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--
Steve Murphy
Digium, Inc. | Software Developer
an expr has arithmetic operators, but no variable
refs, then you get this message.
Yes, I *could* have made it more intelligent. File a bug, and I'll
see if I can do so. At the worst, you can ignore this warning, or
I can simply remove this overly-simple warning.
murf
>
> Thanks,
&
Playback(greetings/fathers);
}
else
{
Playback(greetings/hello); // None of the above? Just
a plain hello will do
}
murf
--
Steve Murphy
Digium, Inc. | Software Developer
57 Lane 17, Cody, WY 82414 USA
direct
_12X
>
> [b](a)
> exten => _1.
>
> Would result in [b] having:
> exten => _12X
> exten => _1.
>
> What's the ael equivalent?
I guess, if you use the #include , you could save the exten =>
_12X in a file snippet and include it where you want it.
I
[EMAIL PROTECTED]
> [mailto:asterisk-users-
> -->> [EMAIL PROTECTED] On Behalf Of Anthony Francis
> -->> Sent: 10 December 2008 18:19
> -->> To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
> -->> Subject: Re: [asterisk-users] CDR Design
>
ce it has of ever getting implemented in
> the near future. Getting a basic accurate CDR system in place does not
> preclude future enhancements but without it they'll just add another
> few layers to the house of cards.
>
> Regards,
>
> Greyman.
>
--
S
sk :). In fact the LEGO paradigm
> would be the ideal solution. I think that asterisk should cope
> with both situations instead of just choosing one.
> I think we all agree on that.
I agree, too. Only trouble is, there's only so many hours in a day!
murf
--
Steve Murphy
r, fetch it to your machine via svn checkout:
http://svn.digium.com/svn/asterisk/team/murf/RFCs
And publish your scathing comments, loathing, compliments, whatever,
and we can hammer out the spec.
murf
--
Steve Murphy
Software Developer
Digium
smime.p7s
Description: S/MIME cryptographic sig
g
and choose the (view) of the first numbered revision in the list to
see the latest version.
murf
--
Steve Murphy
Software Developer
Digium
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27;m in middle yet). So, i hope this will go on and allow me to
> respond with some objective comments.
Atis--
I welcome your input. I don't want to write junk.
And to make this useful to as many users as possible,
I need to know what they nee
t had to be corrected.
Now, it might end up being a walk in the park, and maybe I'm just
being pessimistic, but personally, as a programmer, a CDR spec is
going to be easier to bill from than a sequence of interlaced events.
murf
>
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t; > Regards,
> >
> > Greyman.
> >
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murf
> Any thoughts?
>
> Andrew Thomas
> Technical Services Manager
> DataVox Ltd
> Saddleworth Business Centre
> Huddersfield Road
> Delph, Oldham
> OL3 5DF
>
--
Steve Murphy
Software Developer
Digium
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On Mon, 2008-12-01 at 10:55 -0600, JD wrote:
> Steve Murphy wrote:
> > [...]
>
> I love it! You will have it done later today, correct? (joke.)
>
> Just a non-technical/social suggestion: don't call this CDR. Call it
> "Enhanced CEL" or something like that
ouch the
> generation of the multileg format.
>
> Freddi
Freddi--
Very interesting. Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID set
has been closed. If so, then the entire set is finished. We can use this
On Tue, 2008-11-25 at 08:06 +, Grey Man wrote:
> >On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy <[EMAIL PROTECTED]> wrote:
> > For the moment, let's not worry about the implementation. Let's
> > get consensus on the spec first. In the scenario, where A calls B
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