Re: [Asterisk-Users] X100P connected as extension to Panasonic 616 EASA-PHONE

[Jamie Carl]

So what you're saying is, you can make inbound/outbound calls on 
asterisk using an X100P connected to the PSTN.   Then you unplug the 
X100P from the PSTN and plug it into a Panasonic 616 PABX and suddenly 
nothing works?  Is that correct?


And this is a problem with asterisk because.?

I think you'll find the Panasonic box needs some attention...

Jamie


Guillermo Salas M wrote:


Hi all.

I`ve installed a X100P on my box and is working well with incoming and
outgoing calls as a trunk with one PTSN line.

I want to connect the X100P to my Panasonic 616 EASA-PHONE as an
internal extension to permit to users to make calls to SIP devices from
analog phones, the problem is when I dial the ext number where the X100P
is connected I get busy tone.

What config I need to change to my asterisk files to permit the
commented in the last paragraph?

Best Regards,



Guillermo.

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Jamie Carl
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82 Wentworth Ave
Kingston  ACT  2604

PO Box 4833
KINGSTON ACT  2604

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M 0413 955 956
F 1800 988 000

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RE: [Asterisk-Users] Nasty little incident ...

[Jamie Carl]

:
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Re: [Asterisk-Users] how to make a dialplan on bases of Caller

[Jamie Carl]

Maybe have your uas registered in different contexts for outbound calling.  Then have those contexts only available to dial the appropriate gw?

Just a thought.  I know you could probably do this with wildcard source routing but that seems like overkill.

Jamie



On Tue, 2005-06-14 at 18:09 -0700, Kamran Ahmad wrote:


Hello

i have two GWs and some uas. i want if ua (bw 3000 to
4010) is calling any number then this call will be
routed to first GW and if ua (bw 4020 to 5000) want to
call any number this call will be routed to second GW.


Gateways=GW1,GW2
UAs=3000 to 5000

if 3000 wants to call any number ip or pstn then
Dial(GW1), if 4500 want to call any number ip or pstn
then Dial(GW2)

thanks
Kamran



		
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[Asterisk-Users] Asterisk and grandstream weird call probs

[Jamie Carl]

Hey all.  I've got a weird problem with the grandstream budgetone101 and asterisk that I'm having no luck finding any info on.  I'm positive it's a grandstream problem but i'm hoping someone here can at least point me in the right direction.

Basically, (and it's a simple problem) if a user taps the hook switch quickly they get dialtone again but it does not hangup the existing call.  The user can then make another call, however, i have incominglimit=1 in sip.conf so they cannot.  This means the original call get's lost.  Does anyone know how to retrieve the call?  Or at least where there is some documentation on this 'feature'?

TIA





-- 
Jamie Carl <[EMAIL PROTECTED]>
Resident Geek
Achieve Corp.
+61262648200





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Re: [Asterisk-Users] Supervised/Attended transfers (working, but more Qs)

[Jamie Carl]

Ok, i just got it working with a CVS version.  Cool.  :)  Answered my 
own question.


One more question tho

Will it only work with the keys defined in features.conf or is it 
possible to get it to work using the 'transfer' button on my grandstream 
budgetone101?   I can already hear my users complaining that they'd like 
attended xfers using the transfer button. :(


At the moment i have it mapped to *2 (the default).  Is this possible 
with the budgetones?


Jamie


Jamie wrote:


Hey all,

I've been trying to get supervised transfers working without success.  
I'm currently running 1.0.7-stable and think it might be a version 
problem.  Is the supervised transfer feature available in 1.0.7 or do 
i need to suck down a new version from CVS?


Otherwise, apart from setting up features.conf, is there anything else 
i'm missing?


TIA,

Jamie.






--
Jamie Carl
Resident Geek
Achieve, Corp.

82 Wentworth Ave
Kingston  ACT  2604

PO Box 4833
KINGSTON ACT  2604

T 1300 139 215
M 0413 955 956
F 1800 988 000

W www.achievecorp.com.au

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RE: [Asterisk-Users] Softphone for PocketPC or iPaq

[Jamie Carl]

SJPhone from SJLabs.

www.sjlabs.com

Also, a lot of simple questions like this can be answered by looking at
www.voip-info.org
There is a large section there on different soft/hardware phones.

Regards,

Jamie Carl
Chief 'Stuff' Officer
J-Code International
Email: [EMAIL PROTECTED]
PH: +61414365466
IAXTel: 17004250969



> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Sudhir Kumar
> Sent: Thursday, 23 September 2004 1:03 PM
> To: [EMAIL PROTECTED]
> Cc: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Softphone for PocketPC or iPaq
>
>
> Is there a soft phone for PocketPC or iPaq? If not, is
> someone working on it? I will be more than willing to
> contribute my mite if needed.
>
> Thanks,
> -- sudhir
>
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RE: [Asterisk-Users] Optus Australia Multiline SHDSL service

[Jamie Carl]

What is the type/model of the Adtran box?

I was under the impression the Optus Multinet network (of which
MultiLine is a product of) used 2 boxes onsite.  One an SHDSL NTU and
the other a voice router.  That is, unless things have changed since I
left.

The service has definately been completely installed?  Usually they'd
install the NTU and not put the voice router onsite until sometime
later.

Normally though you would be provided with E1s by Optus which the TE401p
should work fine with.

Regards,

Jamie Carl
Chief 'Stuff' Officer
J-Code International
Email: [EMAIL PROTECTED]
PH: +61414365466
IAXTel: 17004250969



> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> duncan hall
> Sent: Wednesday, 22 September 2004 5:14 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Optus Australia Multiline SHDSL service
>
>
> Hi,
>
> I am currently trying to find a replacement for a dinosaur
> PBX and want
> to replace it with a VoIP solution.
>
> We have just moved our lines over to an Optus Multiline from
> a Telstra
> ISDN Onramp 30 service with 100 lines.
>
> My question for you good people is what sort of hardware do I need to
> interface Asterix into the Optus Multiline? The Optus service is
> terminated in my office to a SHDSL NTU from Adtran and has two RJ45
> conenctors on the back of it. Has anybody tried this yet?
>
> Thanks in advance.
>
> Duncan
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RE: [Asterisk-Users] Zaptel and Linux Distros

[Jamie Carl]

>
>
> Jamie Carl [EMAIL PROTECTED] wrote:
> > Just a quick question.   Are there any known issues using the zaptel
> > drivers on different linux distros?  ie: is a 2.4 kernel the only
> > requirement?
> >
> That should read "2.4 or later".  I'm using the Linux 2.6.8
> kernel with the Gentoo distro.  In theory, the Zaptel driver
> should work on any 2.4 through 2.6-based distro.  I haven't tried 2.7.
>

Excellent.  If it runs on 2.6 I'll be able to give FC2 a shot and see if
that makes any difference.

> >
> > I ask because I have an X100P i'm still trying to get work properly
> > and I'm thinking there is nothing wrong with the card itself.  It's
> > setup correctly as far as I can tell and it used to work when I was
> > using RedHat 8 and 9.  But since I went to Fedora Core 1 I
> don't think
> > it's ever worked properly.
> >
> > Anyone else using Fedora Core 1 and an X100P without issues?  It
> > behaves very strangely and has the same symptoms as setting it to
> > ground start signalling instead of loop start on a loop start PSTN
> > line.  However it is definately set to fxsls and even fxsks
> has been
> > tried.
> >
> I have a X101P at home (using Koolstart) and that performs
> almost acceptably.  I'll be dumping it in favour of a Sipura
> SPA-3000 soon.
>
> I've never used Red Hat Fedora, so I can't comment on that.
> Perhaps you should try FC2 instead of FC1 or, even better, try Gentoo.
>

If FC2 doesn't make any difference I may try rolling back to maybe RH8
or 9 just to confirm if it's a hardware or OS issue.



Thanx

Regards,

Jamie Carl
Chief 'Stuff' Officer
J-Code International
Email: [EMAIL PROTECTED]
PH: +61414365466
IAXTel: 17004250969





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[Asterisk-Users] Zaptel and Linux Distros

[Jamie Carl]

Hey all,
Just a quick question.   Are there any known issues using the zaptel 
drivers on different linux distros?  ie: is a 2.4 kernel the only 
requirement?

I ask because I have an X100P i'm still trying to get work properly and 
I'm thinking there is nothing wrong with the card itself.  It's setup 
correctly as far as I can tell and it used to work when I was using 
RedHat 8 and 9.  But since I went to Fedora Core 1 I don't think it's 
ever worked properly.

Anyone else using Fedora Core 1 and an X100P without issues?  It behaves 
very strangely and has the same symptoms as setting it to ground start 
signalling instead of loop start on a loop start PSTN line.  However it 
is definately set to fxsls and even fxsks has been tried.

I've purchased another X100P just in case it IS a hardware issue.
tia...
--
Regards,
Jamie Carl
Chief 'Stuff' Officer
J-Code International
[EMAIL PROTECTED]
PH: +61414365466
IAXTel: 17004250969
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Re: [Asterisk-Users] Zaptel 'Under the Hood' Project

[Jamie Carl]

Victor Rini wrote:
Hello,
After poking and prodding at Asterisk and Zaptel for over a couple 
years now, I've dedicated some time to actually reading the code and 
trying to figure it out.

It's been fascinating. With the driver source on one part of the 
screen and a pdf of "Linux Device Drivers" on another part I've 
aquainted myself with device driver programming and the interesting 
hardware on the wildcards. I've always thought Asterisk and Zaptel 
were two of the coolest FOSS projects around and now that I've
spelunked through the code a little bit I'm curious:

Has anyone ever wrote a zaptel "under the hood" type of document, 
discussing how the pseudo tdm bus works, the zaptel hardware, etc? If 
so, please point me there.

If not, I'd like to take a stab at compiling a paper or article about 
zaptel for a general audience, technically inclined but not hard core 
technical, i.e. people like me who
have used asterisk but always wondered how it worked down to the 
hardware, spans, channels, chunks, samples level. Some help from the 
community of course would
be great, perhaps through using a blog or wiki.

Once the zaptel "dragon" is dispatched, I'd then focus on Asterisk.
What do you all think?
Regards,
Victor
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I can't speak for anyone else, but I sure as hell would be interested in 
such a document.  I don't think it would even just be for the 
"technically inclined but not hard core technical" guys either.  I 
consider myself pretty "hard core" but I just don't have the time to sit 
down and learn about how it all works on the inside.  There's just too 
many other projects that need to be done.  So in my opinion, a document 
that just lays it out in plain english would save me a heck load of time 
and allow me to learn about something that I unfortunately just don't 
have the time (or motivation) to figure out for myself and therefore 
probably wouldn't end up learning about otherwise. :)

My 2c.
Jamie
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Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

[Jamie Carl]

Bob Knight wrote:
I have MIBs for whatever version I am running that I am more than
happy to share.  Anyone know where I can place these for public access.
Sort of like the freedomphones site for Polycom.  We could then
put pointers on the wiki.
Thanks for the info tho.  If mbrowse is console based it will be very 
useful. :)

It has gui (X, gtk I think) if that is what you mean by console based.
I can ssh into a remote * server and do get walks on my 1204's.

Bob,
I've managed to source the MIBs from another extremely helpful list 
member so hopefully I'm all sorted.  :)

As for posting them, as I'm sure there are others out there that are 
interested, there is a website called www.mibdepot.com which is trying 
to collect as many MIBs as possible and currently has a request for the 
APA III-4FXO MIB.  If you email it to the webmaster of that site he'll 
post it as part of his collection.  I found this site while I was 
looking for it myself so hopefully others will look there too as they 
already have quite a few MIBs available.

Jamie
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Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

[Jamie Carl]

Bob Knight wrote:
There is a linux package called mbrowse that you can use with your 
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not 
spent
that much time on it.

I don't even have the MIBs which is half the problem.  I can do certain 
things using windoze SNMP software, but not exactly being a guru on SNMP 
i'm guessing that without the MIBs i'm pretty much stuffed.

Anyone with MIBs they can send me?  hehe  Please? :)
Thanks for the info tho.  If mbrowse is console based it will be very 
useful. :)

J
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Re: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

[Jamie Carl]

Thanks to everyone for their help and comments on this.  You've all been 
very helpful.  I've actually got outbound calls working on it fine right 
now without having to change the configuration on the Mediatrix box at 
all, as I don't have the Unit Manager Software at the moment.  Outbount 
seems to work well but without inbound it means I can't put it in place 
for general use.  I have my 'reseller' tracking down the software for me 
right now so hopefully he'll be able to find it for me. :)

Asterisk doesn't seem to have any issues working with the APA III-4FXO 
at all as yet. 

Thanks again guys.
J

Gonzalo Gasca Meza wrote:
Here is my configuration for MEdiatrix 1204, by default the 1204
strips one digit, so it is not necessary to use:
To dial OUTSIDE
EXTENSIONS.CONF
[locales]
;ignorepat => 9
exten => _9,1,Dial(SIP/[EMAIL PROTECTED]
)
exten => _9,2,Congestion
exten => _9,102,Congestion
To receive calls
[from-pstn]
;Incoming calls from Mediatrix 1204, the 1204, sends an invite to
[EMAIL PROTECTED] 
exten => ,1,Dial(SIP/100,20)
exten => ,2,Voicemail(u100)
exten => ,102,Voicemail(b100)
exten => ,103,Hangup

***
SIP.CONF
;Mediatrix Telecomm 1204
[Mediatrix]
type=peer
host=110.10.200.10
mask=255.255.255.255
context=from-sip
qualify=yes
canreinvite=yes
disallow=g729
nat = yes
In MEdiatrix 1204 use a program called Unit Manager Network a
Configure the first port as extension  for port 1, in option
SIP. as user agent. also edit registar an dproxy SIP as the IP
address of Asterisk.
Works VERY GOOD with one line, although i have seen some scenarios
with more than 1 line which experince problems.


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[Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?

[Jamie Carl]

Hi all,
I just picked myself up a Mediatrix FXO SIP gateway to play around with 
and hook into Asterisk but have no documentation.

Are there default passwords or IP's that I need to know if I do a 
factory reset? 

Or better still, would anyone have a User Manual they could send my 
way?  Any help would be appreciated.

TIA.
Jamie
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[Asterisk-Users] FXO probs in Aus. Should I give up?

[Jamie Carl]

Hey all,

I've been trying to get my X101P working again as of late (it used to
work great) and before I decide to trash the card I thought I'd post up
my symptoms to see if anyone has any ideas.

My old working config was basically 1 channel running fxsks signalling.
It was working great with no echo, busy detect worked well and I was
very impressed considering this is all off and Australian PSTN line for
which the X101P is not certified.  (hhh).  So one day I update the
zaptel drivers (not sure if this caused it however), and now it cannot
go off-hook on it's own.

Outbound Symptoms are:

Placing a call from a SIP softphone, * will cease the zap channel and
look like it's working, but no audio can be heard (ring tone, etc) on
the softphone. Now, if I go off-hook on a POTS phone running parallel to
the X101p suddenly everything comes to life.  If I go off-hook on the
parallel phone before the X101p tries to dial, everything works fine.
But on it's own, it's a no go.

Inbound Symptoms:

The zap channel detects ring, ceases the channel and begins normal call
flow and in my test setup going straight to voicemail.  The caller can
hear the call is answered but again, no audio.  Going off-hook again on
the parallel phone kicks everything back into life.

Now here's the kicker.  I have an old frame-relay voice switch I
'borrowed' from an ex-employer and have configured some slots for FXS to
run back-to-back with the X101p.  It works first time, every time.  Only
difference I can think of between them is that the voice switch is from
the US and therefore uses US tones, etc.  ??

I have tried both Loopstart and Koolstart signalling.  Groundstart will
not load when I use 'ztcfg' for some reason.  So is there something I'm
missing.  This used to work fine.  Has something changed in the zaptel
driver? Are there any undocumented settings I can tweak to possible get
this working again?

I'm about to chuck the card and go for a SIP or MGCP gateway but if I
can not spend the cash, I will.  Anyone with ideas?

Thanks heaps.

Regards,

Jamie Carl
Chief 'Stuff' Officer
J-Code
Web:http://www.j-code.net
Email:  [EMAIL PROTECTED]



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RE: [Asterisk-Users] FXOs

[Jamie Carl]

The only FXO interface that I have at the moment is an X101p.  It was
working great up until about a year ago and then something weird
happened and I haven't used it since (until recently).  Now it would
seem it just doesn't like my PSTN line however it works fine running
back-to-back on another non-PSTN FXS interface.  Still working on it tho
so I may get it working again soon.




Regards,

Jamie Carl
Chief 'Stuff' Officer
J-Code
Web:http://www.j-code.net
Email:  [EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, 28 August 2004 1:41 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] FXOs


Hi All,

I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.

I also took part in Sipura's beta program, for the SPA-3000. While it
can be an improvement over the X100p, it presently has echo problems
that make it unusable. Sipura has not acknowledged the problem ( at
least to me) although several in the user community make refernce to new
firmware that might address the issue, real soon now.

I see a lot of activity recently on-list about the TDM-400. Of course,
mentions on-list are more than likely the result of people having
problems. We don't hear about people who have no issues with a product.

So, the nature of my inquiry is to explore how many people out here have
good/great experiences with the various small FXO adapters? While the
TDM-400 is my next possible purchase I'd also like to hear about devices
from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so
many products being offered I would hope that we have some collective
experience with each one.

Thanks,
Michael



Michael Graves
Sr Product Specialist
Pixel Power Inc
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RE: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*


Has been fixed..   This is what happens when you don't pay your bills..
:(

J

> -Original Message-
> From: PJ Welsh [mailto:[EMAIL PROTECTED]
> Sent: Friday, 3 October 2003 12:52 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] CDR Web Search Frontend
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> On Thu, Oct 02, 2003 at 09:58:37AM +1000, Jamie Carl wrote:
> ...
> > As for the rest of this discussion, I have already started
> > work on this Asterisk Web Interface. (visit
> > http://astweb.sourceforge.net).  The current release is
> > still only the CDR section, but things are starting to
> > evolve and I expect to have something usable in the next
> > few weeks.  It is being written in PHP and will attempt to
> > use ZERO OS-DEPENDANT code.
> ...
>
> Sorry, preview selection generates:
>
>  Your DNS2Go account has been disabled.
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

Ok, is it time for my comments on all of this?

On Wed, 01 Oct 2003 16:56:22 -0500
 Steven Critchfield <[EMAIL PROTECTED]> wrote:
On Wed, 2003-10-01 at 16:29, CW_ASN wrote:
Yes, I know... but some people uses Windows and hates 
web frontends (some
customers, for example)...
I hate Windows platforms, but I'm only a technician...

If this is the case, obviously you need more experience in 
creating user friendly web frontends.  Some that I have 
seen are brilliant and look great on any OS.  PHP is best 
suited to what we are trying to do, because it's powerful, 
it has all the feature we want to use and it runs on the 
server.  Introducing FTP transfers and other such 
mechanisms is just asking for trouble.  Plus tying it to a 
windows system is very Anti-productive.

Then please reread the whole of my argument. I was 
arguing only against
a front end that couldn't be used on other platforms 
also. This is why I
was so down on VB, it is only usable on Windows. I was 
mildly down on
.net stuff only because of the potential to have the mono 
rug ripped out
from under you. At least with a C++ app and a cross 
platform widget set,
you could write for windows and get linux and BSD for 
nearly free.

Steven, i resent the discrimination of VB programmers, 
especially the comment: "those types of programmers are 
unlikely to use linux in any fashion".  I've been writing 
in VB for 8 years and it was the first language I learnt. 
I still use it occasionally.  Even though I use C on both 
windows and Linux, Python and now PHP.  It has it's uses. 
:)  Agreed though that this admin project is not one of 
them. :)

As for the rest of this discussion, I have already started 
work on this Asterisk Web Interface. (visit 
http://astweb.sourceforge.net).  The current release is 
still only the CDR section, but things are starting to 
evolve and I expect to have something usable in the next 
few weeks.  It is being written in PHP and will attempt to 
use ZERO OS-DEPENDANT code.

Regards,

Jamie Carl
Jazz Inc.
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Web:www.jazz-inc.net
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

On Tue, 30 Sep 2003 00:02:06 -0700
 "Paul Crick" <[EMAIL PROTECTED]> wrote:
  itemised billing from a telco, User Accessable)
Sounds good. Would you want to extend it any, using place 
names etc? So
calls to +1604xxx show up as "Canada" or "BC, Canada" (or 
even "Vancouver
BC, Canada") or am I just being a bit too ambitious (and 
annoying) there?
Yes ambitious, No, not annoying, and my comment is "oh my 
god, who's going to collate a list of Names/Area-codes?? 
Stuffed if I'm doing it. :)  

;-)

* Asterisk Management (initially based on PHPconfig)
Can you elaborate on this a bit? Do you mean holding all 
the config
parameters in databases which are either accessed 
natively or used to
generate text config files (as many web hosting control 
panels do)? Or do
you just mean including the existing web config stuff in 
to the bigger grand
scheme, more or less as-is?

Personally, I havn't thought of integrating PHPConfig into 
this.  I haven't really had a look at it either(i'm more 
of a CLI guy), so I don't know what parts of it would be 
useful for what we are trying to achieve.

I like the ideas of user self service.. or user filtered 
access to the CDR
stuff?

Will there/should there be levels of user access and 
stuff to restrict
access to certain areas?

Yes, user access restrictions are in mind.  Don't know how 
we're going to do this yet, perhaps rape the agents.conf, 
manager.conf or other pre-existing Asterisk conf file for 
this info.  Otherwise, storing our own, or using .htpasswd 
files mite be the go.

Big can of worms we're opening, but I'm up for some PHP 
coding, I have spare
cycles right now.. and with a snazzy front end, it's more 
value to
Asterisk..
Mmm..  I like eating worms.  I ate lots as a kid. :)
Email me off-list if you're interested in joining the 
jazz-wagon. :)  [EMAIL PROTECTED]

Regards,

Jamie Carl
Jazz Inc.
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

On Tue, 30 Sep 2003 07:43:34 +0100
 WipeOut <[EMAIL PROTECTED]> wrote:
Thats COOL!!!. :)

Is GPL the correct licence for it??

I am not so hot on all that licencing stuff and all I 
hear is that licence X is not compatible with licence Y 
and so this code needs to removed and yada-yada-yada 
blah-blah-blah..

Would it not be better to use LGPL (which I believe is 
less restrictive) or X11/MIT Licence which has very few 
limitations as I see it..

Later..
GPL should be sufficient.  I went over it during my lunch 
break.  Seems like the users have more rights than the 
Authors. :)  Basically what it comes down to is that 
anyone can use/modify it in any way, as long as they 
release it GPL as well.  I found it to be quite lax on 
restrictions for use/distribution to be honest.

It also means that if it's modified, I can't be blamed for 
anything, which is all I'm worried about.  Given that it 
is a separate entity to Asterisk, it should not use LGPL 
as it is not a library.

J

Regards,

Jamie Carl
Jazz Inc.
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

All registered with Sourceforge.  The new project will be 
called 'AstWeb' and will be available under the GNU 
General Public Licence.  The current release of AstCDR 
will be ported over in the next day or so along with the 
addition of a few features and labelled 'AstWeb v0.4'.

The project is currently pending approval with SF.net and 
once approved will be available for download from the 
SF.net website.

Thanx all,

J

On Mon, 29 Sep 2003 23:57:23 -0400
 Leif Madsen <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Jamie Carl wrote:

I think we're getting away from the original purpose of 
this program.  
Are people really that desparate for a full, web-based 
admin/user 
interface?

If so, tell me, and we'll make this an official project 
rather than just 
some code I slapped together in an hour. :)
I think it's fair and safe to say they are...

Thanks,
Leif Madsen.
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

On Mon, 29 Sep 2003 22:08:28 -0400
 "Uriel Carrasquilla" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
How about including VoiceMail viewer/retriever.
URiel

I think we're getting away from the original purpose of 
this program.  Are people really that desparate for a 
full, web-based admin/user interface?

If so, tell me, and we'll make this an official project 
rather than just some code I slapped together in an hour. 
:)

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
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Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

Guys!  I'm putting the source up on SourceForge on my 
existing account.  Questions is this tho:

What should we call it?

Is AstCDR good/fancy enough?  

Suggestions please!  I would like to get this on SF by the 
end of the day. (it's 9:33am here).

Someone I know once said:
 "I'm not a coder, I'm just an idea man."
Well, I'm not a marketing man, just a coder.

:)

J

On Tue, 30 Sep 2003 00:06:35 +0100
 WipeOut <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Brian West wrote:

I have found the CDR in general to be a problem, We use a 
system that
allows a user to simply click a phone number in a web 
page and then PHP
drops a call file into the /outgoing directory.. These 
calls are not
logged at all.. not in the text file or the MySQL..
  

Can I make calls thru your system please?  Sounds like a 
major issue that
needs to be fixed ASAP mmmkay!


Sorry Dude, The app is only available internally and the 
SIP phone that Asterisk routes the call to is hardcoded 
to the user account.. :)

But yes it is annoying that its not logged becasue its 
messing up out sales people's call stats.. So we can't 
give them the "evil eye" when the call rates are low 
bacasue we have no idea how many calls thay are actually 
making..

Just as well we are not trying to do any billing!! Man 
would we be loosing money..

Later..

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RE: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*
> Jamie Carl wrote:
>
> >*This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> >
> >
> >Don't spose I get any say in this??
> >:)
> >
> >J
> >
> >
> >
> Sure you do, If you have a CVS server and want the
> development to run on
> it then thats also cool.. :)
>
> Thats the beauty of community development.. Everyone has a say..
>
> Later..

Even the guy that wrote the damn thing?  Kewl..  I love this open source
community stuff.
:)
hehe

My thought are, go with sourceforge.  Mainly cause I already have an
account set up on it, plus it's always the first place I look for stuff,
so I'm sure it's the first place others look for stuff too.

J



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RE: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*


Don't spose I get any say in this??
:)

J


> -Original Message-
> From: WipeOut [mailto:[EMAIL PROTECTED]
> Sent: Monday, 29 September 2003 8:09 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] CDR Web Search Frontend
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
>
> >I was thinking of using
> >
> >http://developer.berlios.de/
> >
> >As SF has had many problems recently :(
> >
> >Regards
> >
> >Mark
> >
> >
> >
> Yea, I have noticed Sourceforge has been a little flaky
> lately.. Thought
> they would have been on top of it quicker..
>
> Later..
>
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RE: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*


It's in the archives.  People on this list usually don't take too well
to repeating stuff. :)
(i'm not fussed tho)

http://asterisk.jazz-inc.net

Yes, the source is of course available for download.
:)

Enjoy!

J

> -Original Message-
> From: Uriel Carrasquilla [mailto:[EMAIL PROTECTED]
> Sent: Sunday, 28 September 2003 1:05 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] CDR Web Search Frontend
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> does it include the source in PhP?
> what was the link again please?
> Uriel
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Jamie Carl
> Sent: Saturday, September 27, 2003 3:42 AM
> To: Asterisk Users (E-mail); Asterisk Dev (E-mail)
> Subject: RE: [Asterisk-Users] CDR Web Search Frontend
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
>
> Hey all,
>
> New versions available.  Now written in PHP with totals for Billing
> Seconds and Duration.
>
> Help yourselves and please send me more suggestions!!!
> Thanx!
>
> J
>
> > -Original Message-
> > From: Dimitri Bellini [mailto:[EMAIL PROTECTED]
> > Sent: Friday, 26 September 2003 10:40 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] CDR Web Search Frontend
> >
> >
> > *This message was transferred with a trial version of
> > CommuniGate(tm) Pro*
> > Hi Carl
> > i see web frontend i action is very good!! The total
> > time at end is good
> > thing.
> > Thanks for great work. Can you put the script in some place
> > to download.
> >
> > Dimitri
> >
> > > *This message was transferred with a trial version of
> > CommuniGate(tm) Pro*
> > >
> > > Hey all,
> > >
> > > I've just done a quick (but functional) web front end for
> > searching the
> > > CDRs in a MySQL database.  Anyone interested in trying it
> out?  I'm
> > > wondering what to add to it next.
> > >
> > > So far you can seach using source, destination, CLI,
> > channel and date
> > > ranges.  It also displays ALL fields in the database table.
> > >
> > > If interested, email me on [EMAIL PROTECTED]  Do not reply
> > directly to
> > > this email, it will bounce.  Depending on the level of
> > interest, I may
> > > post this somewhere for your free downloading pleasure.
> > >
> > > Regards,
> > >
> > > Jamie Carl
> > > Jazz Inc.
> > > http://www.jazz-inc.net
> > > Email: [EMAIL PROTECTED]
> > > JID: [EMAIL PROTECTED]
> > > Phone: +61-414-365466
> > >
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
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>
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*
Gimmie a break, I only learnt PHP yesterday..
:)

J

- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 27, 2003 10:13 PM
Subject: Re: [Asterisk-Users] CDR Web Search Frontend


> *This message was transferred with a trial version of CommuniGate(tm) Pro*
> > Since * and MySQL have had a licensing scuffle, is there a way to set it
> > up so that we can specify wether or not it's in the mysql database, or
> > use the plaintext file that * generates with cdr_csv.so?
>
> Or do something really smart like the Perl guys and have a
> backend-mostly-independent DB infrastructure.  Hell I think that PHP
> finally smartened up and went this way, too.
>
> Regards,
> Andrew
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RE: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*

Hey all,

New versions available.  Now written in PHP with totals for Billing
Seconds and Duration.

Help yourselves and please send me more suggestions!!!
Thanx!

J

> -Original Message-
> From: Dimitri Bellini [mailto:[EMAIL PROTECTED]
> Sent: Friday, 26 September 2003 10:40 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] CDR Web Search Frontend
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> Hi Carl
>   i see web frontend i action is very good!! The total
> time at end is good
> thing.
> Thanks for great work. Can you put the script in some place
> to download.
>
> Dimitri
>
> > *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> >
> > Hey all,
> >
> > I've just done a quick (but functional) web front end for
> searching the
> > CDRs in a MySQL database.  Anyone interested in trying it out?  I'm
> > wondering what to add to it next.
> >
> > So far you can seach using source, destination, CLI,
> channel and date
> > ranges.  It also displays ALL fields in the database table.
> >
> > If interested, email me on [EMAIL PROTECTED]  Do not reply
> directly to
> > this email, it will bounce.  Depending on the level of
> interest, I may
> > post this somewhere for your free downloading pleasure.
> >
> > Regards,
> >
> > Jamie Carl
> > Jazz Inc.
> > http://www.jazz-inc.net
> > Email: [EMAIL PROTECTED]
> > JID: [EMAIL PROTECTED]
> > Phone: +61-414-365466
> >
> >
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

Good suggestion!  Duely noted.  Check after the weekend 
and it'll be there.  

(i would do it tonite, but it's friday nite and there is 
alcohol to be consumed)

J

On Thu, 25 Sep 2003 22:36:31 -0400
 "Andrew Joakimsen" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Looks great. One suggestion would be to add a total at 
the end with
total/billable durations and total number of calls.


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jamie Carl
Sent: Thursday, September 25, 2003 10:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CDR Web Search Frontend

Ok, wow.  Didn't expect as many responses as I got.
 Didn't think this would spark so much interest.
Anywayz.  Some of the questions asked will be answered
here, rather than emailing everyone individually with 
the
same answers.

Currently, the backend is written in Python2 with the
MySQLdb module.  However, due to some interest I will be
porting it to PHP.
Also, if you want to see it in action, you can look at
http://asterisk.jazz-inc.net to see my working (but 
still
under development) version.

I am going to put links on this page to packaged source
for free download.  Just have to sort out disclaimers 
and
all that so you all cant sue me. :)

Thanx guyz,

J

On Fri, 26 Sep 2003 03:22:45 +1000
  "Jamie Carl" <[EMAIL PROTECTED]> wrote:
>
>Hey all,
>
>I've just done a quick (but functional) web front end 
for
>searching the
>CDRs in a MySQL database.  Anyone interested in trying 
it
>out?  I'm
>wondering what to add to it next.
>
>So far you can seach using source, destination, CLI,
>channel and date
>ranges.  It also displays ALL fields in the database
>table.
>
>If interested, email me on [EMAIL PROTECTED]  Do not
>reply directly to
>this email, it will bounce.  Depending on the level of
>interest, I may
>post this somewhere for your free downloading pleasure.
>
>Regards,
>
>Jamie Carl
>Jazz Inc.
>http://www.jazz-inc.net
>Email: [EMAIL PROTECTED]
>JID: [EMAIL PROTECTED]
>Phone: +61-414-365466
>
>
>
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
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Re: [Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

Ok, wow.  Didn't expect as many responses as I got. 
Didn't think this would spark so much interest.

Anywayz.  Some of the questions asked will be answered 
here, rather than emailing everyone individually with the 
same answers.

Currently, the backend is written in Python2 with the 
MySQLdb module.  However, due to some interest I will be 
porting it to PHP.

Also, if you want to see it in action, you can look at 
http://asterisk.jazz-inc.net to see my working (but still 
under development) version.

I am going to put links on this page to packaged source 
for free download.  Just have to sort out disclaimers and 
all that so you all cant sue me. :)

Thanx guyz,

J

On Fri, 26 Sep 2003 03:22:45 +1000
 "Jamie Carl" <[EMAIL PROTECTED]> wrote:
Hey all,

I've just done a quick (but functional) web front end for 
searching the
CDRs in a MySQL database.  Anyone interested in trying it 
out?  I'm
wondering what to add to it next.

So far you can seach using source, destination, CLI, 
channel and date
ranges.  It also displays ALL fields in the database 
table.

If interested, email me on [EMAIL PROTECTED]  Do not 
reply directly to
this email, it will bounce.  Depending on the level of 
interest, I may
post this somewhere for your free downloading pleasure.

Regards,

Jamie Carl
Jazz Inc.
http://www.jazz-inc.net
Email: [EMAIL PROTECTED]
JID: [EMAIL PROTECTED]
Phone: +61-414-365466


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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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[Asterisk-Users] CDR Web Search Frontend

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*

Hey all,

I've just done a quick (but functional) web front end for searching the
CDRs in a MySQL database.  Anyone interested in trying it out?  I'm
wondering what to add to it next.

So far you can seach using source, destination, CLI, channel and date
ranges.  It also displays ALL fields in the database table.

If interested, email me on [EMAIL PROTECTED]  Do not reply directly to
this email, it will bounce.  Depending on the level of interest, I may
post this somewhere for your free downloading pleasure.

Regards,

Jamie Carl
Jazz Inc.
http://www.jazz-inc.net
Email: [EMAIL PROTECTED]
JID: [EMAIL PROTECTED]
Phone: +61-414-365466



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RE: [Asterisk-Users] Does SIP work?

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*



Lack of documentation?


Welcome to the bleeding edge...


Enjoy..

J

> -Original Message-
> From: Tais M. Hansen [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, 24 September 2003 10:54 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Does SIP work?
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Wednesday 24 September 2003 14:38, Andrew Kohlsmith wrote:
> > > Now that I've been unable to register 2 hardware SIP
> phones and one
> > > software (Kphone), I'm beginning to doubt that chan_sip
> works at all.
> > I use SIP to talk to Grandstream 100s every day, and also to the FWD
> > network without issue.  Are you trying to access SIP across
> NAT or other
> > restrictive firewall or something?
>
> No firewall, no nat. Only lack of documentation for
> sip.conf... I got it to
> work now.
>
> - --
> Regards,
> Tais M. Hansen
> ComX Networks
> Tel: +45-70257474
> Fax: +45-70257374
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.2 (GNU/Linux)
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> ILS+9IptUCB6HrsDDLVMrCA=
> =lRx5
> -END PGP SIGNATURE-
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RE: [Asterisk-Users] Does SIP work?

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*

SIP Works fine.  I use it every day.

Check your config.  What errors are you getting when you endpoints try
to register?

Also, go through the mailing list archives as there are sample configs
in there somewhere.

J

> -Original Message-
> From: Tais M. Hansen [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, 24 September 2003 8:25 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Does SIP work?
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Now that I've been unable to register 2 hardware SIP phones
> and one software
> (Kphone), I'm beginning to doubt that chan_sip works at all.
>
> - --
> Regards,
> Tais M. Hansen
> ComX Networks
> Tel: +45-70257474
> Fax: +45-70257374
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.2 (GNU/Linux)
>
> iD8DBQE/cXDg2TEAILET3McRAlWgAJ4/2Y21JU5VkfrO2CMLAMAfOdiszACgk/Yf
> lCKOXryUGv1nQOevTry+rqc=
> =pJJR
> -END PGP SIGNATURE-
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Re: [Asterisk-Users] SIP registration between *'s

[Jamie Carl]

Why?

Use IAX2, it is s much better...

J

On Fri, 19 Sep 2003 11:54:23 +0200
 "Xisco" <[EMAIL PROTECTED]> wrote:
Hi everybody,

I'm trying to SIP register between two asterisk, each one 
have a Public IP. Asterisk told me that Unathorizae

In * one sip.conf

register =>usuario1:pass1@

In * two sip.conf

[usuario1]
type=friend
username=usuario1
secret=pass1
host=
dtmfmode=inband
Logs in * are the followings

In * one logs:

Sip read: >
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
:5060;branch=z9hG4bK488fe503;received=
From: >;tag=as504a35d0
To: >;tag=as2a0e47ce
Call-ID: 
77064d8f2fdfe4746d509dc2488fe503@
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Contact: >
Content-Length: 0

9 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:SIP/2.0
Via: SIP/2.0/UDP 
:5060;branch=z9hG4bK59f913b2
From: >;tag=as4f879ac7
To: >
Call-ID: 
77064d8f2fdfe4746d509dc2488fe503@
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: >
Event: registration
Content-length: 0

 (no NAT) to:5060
Sip read: >
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
:5060;branch=z9hG4bK59f913b2;received=
From: >;tag=as4f879ac7
To: >;tag=as13445743
Call-ID: 
77064d8f2fdfe4746d509dc2488fe503@
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Contact: >
Content-Length: 0

In * two logs:

NOTICE[81926]: File chan_sip.c, Line 4816 
(handle_request): Registration from 
'>' failed for ''

Sip read:
REGISTER sip:SIP/2.0
Via: SIP/2.0/UDP 
:5060;branch=z9hG4bK0f194106
From: >;tag=as35957f60
To: >
Call-ID: 
77064d8f2fdfe4746d509dc2488fe503@
CSeq: 119 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: >
Event: registration
Content-length: 0

11 headers, 0 lines
Using latest request as basis request
Sending to : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
:5060;branch=z9hG4bK0f194106;received=
From: >;tag=as35957f60
To: >;tag=as1538b8a6
Call-ID: 
77064d8f2fdfe4746d509dc2488fe503@
CSeq: 119 REGISTER
User-Agent: Asterisk PBX
Contact: >
Content-Length: 0
Any idea to fix the problem Any special configuration 
in sip.conf

Thanks a lot.

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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[Asterisk-Users] # during ringing causes Asterisk to crash!

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*

Hey all,

Just noticed something that might be an issue.  I have just made
asterisk crash consistently by doing the following.

I have a D-Link DG1102s running MGCP into asterisk and an extension *9
setup which dumps me into my inbound context to simulate calls coming in
from my X100P.  This usually works with no hassles at all.  Except this
time I pressed # too early, as in before it actually answered and the
MGCP IAD was still providing ring tone, and Asterisk just stopped
responding to everything!  I had to do a send a SIGKILL just to kill the
running process.

Here's a quick log just to illustrate what I mean:

  == Spawn extension (default, 121, 1) exited non-zero on
'MGCP/aaln/[EMAIL PROTECTED]'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '*'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '9'
-- Executing Goto("MGCP/aaln/[EMAIL PROTECTED]", "inbound|s|1") in new
stack
-- Goto (inbound,s,1)
-- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", "MGCP/aaln/[EMAIL PROTECTED]|30")
in new stack
-- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: -1, dnd: 0, so: 1, sno: 0
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '#'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'

And at this point everything stops responding.

I have a feeling people will be able to recreate this without any issues
and that the problem lies in the MGCP channel driver.

Anyone else experiencing this?

Regards,

Jamie Carl
Jazz Inc.
http://www.jazz-inc.net
Email: [EMAIL PROTECTED]
JID: [EMAIL PROTECTED]
Phone: +61-414-365466



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RE: [Asterisk-Users] Incoming calls from IAXTEL over NAT

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*

Yup, seen something like that, but I don't have any iax user/friends
setup anywayz.  The only thing I use IAX for it outbound calls to
IAXTEL.

I've analysed this at a lower level and I can't actually see any IAX
packets coming in on my network interface to even start worrying if it's
an incoming config issue.

Next...


p.s. thanx anywayz..

Jamie

> -Original Message-
> From: Paul Cheng [mailto:[EMAIL PROTECTED]
> Sent: Friday, 12 September 2003 4:00 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Incoming calls from IAXTEL over NAT
>
>
> *This message was transferred with a trial version of
> CommuniGate(tm) Pro*
> Don't know if this will help, but have you seen:
>
> http://www.junghanns.net/asterisk/page12.html
>
> ?
>
> On Friday, September 12, 2003, at 03:06  AM, Jamie Carl wrote:
>
> >
> > Hey all,
> >
> > I was playing around with IAXTEL last nite and have outgoing calls
> > working a treat.  I'm sure I woke a few people up in the US with my
> > annoying test calls. :)
> > Anywayz, incoming calls are a different matter.  I have a
> NAT firewall
> > my * box is sitting behind and the server 'appears' to have
> registered
> > correctly with IAXTEL.  Thing is, when I try and call my
> 1700 number
> > to 'loop' back to myself, I get no incoming requests.
> > I've also tried calling the 1700 number from FWD using
> X-Lite (which
> > works to other numbers) and I get no incoming requests
> either.  Just
> > Marks voice telling me the person i'm calling is either
> unregistered
> > or unavailable.
> >
> > Is there anything I need to do to my firewall to get
> incoming calls to
> > work with IAXTEL?  Or should it just 'work'.  I've also tried
> > forwarding my IAX port (5036) to my * box without success.
> >
> > Anyone done something similar that can point me in the
> right direction?
> >
> > Thanx...
> >
> > Regards,
> >
> > Jamie Carl
> > Jazz Inc.
> > Email:  [EMAIL PROTECTED]
> > Web:www.jazz-inc.net
> > Phone:  +61-414-365-466
> > Jabber: [EMAIL PROTECTED]
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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[Asterisk-Users] Incoming calls from IAXTEL over NAT

[Jamie Carl]

Hey all,

I was playing around with IAXTEL last nite and have 
outgoing calls working a treat.  I'm sure I woke a few 
people up in the US with my annoying test calls. :)  

Anywayz, incoming calls are a different matter.  I have a 
NAT firewall my * box is sitting behind and the server 
'appears' to have registered correctly with IAXTEL.  Thing 
is, when I try and call my 1700 number to 'loop' back to 
myself, I get no incoming requests.  

I've also tried calling the 1700 number from FWD using 
X-Lite (which works to other numbers) and I get no 
incoming requests either.  Just Marks voice telling me the 
person i'm calling is either unregistered or unavailable.

Is there anything I need to do to my firewall to get 
incoming calls to work with IAXTEL?  Or should it just 
'work'.  I've also tried forwarding my IAX port (5036) to 
my * box without success.

Anyone done something similar that can point me in the 
right direction?

Thanx...

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] DLink DG-104S

[Jamie Carl]

Works great for me.  I use it everyday as I have a 
DLINK-DG1102S at home hooked up to Asterisk with an X100P.

J

On Mon, 8 Sep 2003 23:33:47 -0400
 "Andrew Joakimsen" <[EMAIL PROTECTED]> wrote:
Actually the one I was thinking of was the DVG-1120 that 
supports SIP.
Or is Asterisk's MGCP protocol working well?


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jamie Carl
Sent: Monday, September 08, 2003 11:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DLink DG-104S

pamAssassin 2.55 (1.174.2.19-2003-05-19-exp)

Yup, they work great..  What country?

U can go to DLINK direct or if in Australia you can talk
to Nick at Salient Networks on 0291442622 or visit
http://www.salientnetworks.com.au
Keep in mind tho these things are about $400 in 
Australia.

J

On Mon, 8 Sep 2003 22:12:52 -0400
  "Andrew Joakimsen" <[EMAIL PROTECTED]> wrote:
>*This message was transferred with a trial version of
>CommuniGate(tm) Pro*
>Does anyone have a source where these can be purchased?
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>>[mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Serge Mankovski
>> Sent: Monday, September 08, 2003 10:02 PM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] DLink DG-104S
>>
>> Hi
>> Did anyone try to setup DLink DG-104S VoIP  Gateway 
with
>>asterisk?
>>
>> Thanks
>> Serge
>>
>> 
_
>> MSN 8 helps eliminate e-mail viruses. Get 2 months
>>FREE*.
>> http://join.msn.com/?page=features/virus
>>
>> ___
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>> 
http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
___
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] DLink DG-104S

[Jamie Carl]

GRAB THEM!

These things are definately overpriced.  But work rather 
well regardless.  That price will probably go up too. 
Items like that will have most of their bids done in the 
last 30 minutes.

J

On Tue, 09 Sep 2003 03:11:59 +
 "Serge Mankovski" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Thank you Jamie.
I am in Canada. There  are used units on Ebay at about 
$60 -$80 USD a piece.

Serge


From: "Jamie Carl" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DLink DG-104S
Date: Tue, 09 Sep 2003 13:03:43 +1000
Yup, they work great..  What country?

U can go to DLINK direct or if in Australia you can talk 
to Nick at Salient Networks on 0291442622 or visit 
http://www.salientnetworks.com.au

Keep in mind tho these things are about $400 in 
Australia.

J

On Mon, 8 Sep 2003 22:12:52 -0400
"Andrew Joakimsen" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Does anyone have a source where these can be purchased?

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Serge Mankovski
Sent: Monday, September 08, 2003 10:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] DLink DG-104S

Hi
Did anyone try to setup DLink DG-104S VoIP  Gateway with 
asterisk?

Thanks
Serge
_
MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*.
http://join.msn.com/?page=features/virus
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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http://join.msn.com/?page=features/virus

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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] DLink DG-104S

[Jamie Carl]

Yup, they work great..  What country?

U can go to DLINK direct or if in Australia you can talk 
to Nick at Salient Networks on 0291442622 or visit 
http://www.salientnetworks.com.au

Keep in mind tho these things are about $400 in Australia.

J

On Mon, 8 Sep 2003 22:12:52 -0400
 "Andrew Joakimsen" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Does anyone have a source where these can be purchased?

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Serge Mankovski
Sent: Monday, September 08, 2003 10:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] DLink DG-104S

Hi
Did anyone try to setup DLink DG-104S VoIP  Gateway with 
asterisk?

Thanks
Serge
_
MSN 8 helps eliminate e-mail viruses. Get 2 months 
FREE*.
http://join.msn.com/?page=features/virus

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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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[Asterisk-Users] Browsing CVS Online

[Jamie Carl]

Hey all,

Does anyone know if there is a way to browse the CVS tree 
online, via a webpage?

This sounds like a stupid request, but I don't have access 
to a shell at the moment and wanted to go over a few 
things.

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP on TCP

[Jamie Carl]

On Wed, 3 Sep 2003 15:47:35 -0600
 Timothy Soos <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
On Wednesday 03 September 2003 09:32 am, Master Abi 
wrote:
Hi

I read through the archives but could not find much 
reference to * using
SIP on TCP instead of UDP for signalling. Can * be 
configured and if so
how. My service provider will only accept SIP signalling 
on TCP.

Thanks

Master
This suggestion is not exactly elegant, yet it may be of 
help:

"The UDP over TCP tunnel is a simple UDP-over-TCP 
application written in Java, 
and is designed so that people behind a firewall can 
connect through a TCP 
connection to play Quake and other UDP-based games."

It is available at:
http://freshmeat.net/projects/udptunnel/?topic_id=907%2C150%2C151
--
This wouldn't exactly help his situation.  Would this 
tunnel require this software running at both ends?  What 
he's talking about is SIP on TCP. Not SIP on UDP on TCP. 

SIP, by design, is supposed to be used with UDP as it has 
a lot of it's own transport control mechanisms, ie, 
retries, acks, etc, which allow for UDPs lack of 
reliability.

However, SIP can run on TCP with no major changes apart 
from altering the stack to establishing a TCP connection 
before signalling.

From section 18 in RFC3261:


All SIP elements MUST implement UDP and TCP.  SIP elements 
MAY implement other protocols.

Making TCP mandatory for the UA is a substantial change 
from RFC 2543.  It has arisen out of the need to handle 
larger messages, which MUST use TCP, as discussed below. 
Thus, even if an element never sends large messages, it 
may receive one and needs to be able to handle them.


So if * wants to be RFC 3261 complient (which to get in 
anywhere commercial, it'll need to be) TCP MUST be 
implemented.

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Sip Software from Nero Folk?

[Jamie Carl]

On 01 Sep 2003 23:23:53 -0500
 Steven Critchfield <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
On Mon, 2003-09-01 at 22:23, Dave Packham wrote:
http://www.nero.com/us/631911127302064.html

Have you all seen this?  

Its a SIP softphone put out by the people that do the CD 
burning software Nero...

Check it out  it works with * 
And the benefit of using a commercial software that costs 
money is ? 

I just love the fact that they claim stunning sound 
quality when all the
variables are outside it's control. Software doesn't make 
you sound card
better. The codecs are standard and therefore not going 
to be improved
by this software. 

Not to mention it only runs on windows, and it's minimum 
requirements is
higher than that needed for an asterisk system. 
--
Steven Critchfield <[EMAIL PROTECTED]>

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C'mon Steven, give the poor Nero folk a break.  I'm sure 
they have marketing drones that they have to keep busy 
coming up with this 'stunning sound quality' crap.

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Dialed Number Identification in analoghunt group

[Jamie Carl]

On Tue, 26 Aug 2003 17:48:55 -0500
 Don Pobanz <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
On Tuesday, August 26, 2003 4:55 PM, Stephen R. Besch 
[SMTP:[EMAIL PROTECTED] wrote:
Does anyone out there know if it is possible to discover 
the dialed
number when a line in an analog hunt group rings?  I 
can't get a
straight answer from our IT folks. We have a 5ess switch 
delivering 4
analog lines which are in a simple hunt group servicing 
our lab.  I
would like to have a different call attendant based on 
which number 
is
dialed so that I can route the calls to the appropriate 
group.  I 
know
that Asterisk can easily do this once I have the 
information to pass
into the dial plan.  The problem is getting the 
information.  While I
know that this is possible with T1, it is, 
unfortunately, a bit
overkill
for 4 lines. Anyone have any suggestions?
If they are pots lines in a hunt group, you won't be able 
to.

If they are analog DID trunks then the dialed number 
would be passed.

My guess is you have pots lines and there is no way to 
find out the 
dialed number.

Don Pobanz

Again, not near my asterisk box so I can't check this out, 
but is it possible to have the different ports drop into * 
in a different context for each line?  That way you could 
just set up an 's' extension in that context for the 
different attendants.

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP vs SCCP vs XML

[Jamie Carl]

On Mon, 25 Aug 2003 18:40:59 -0500
 Adam Roach <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
I'll start by mentioning that the newer Cisco SIP dumps
let you hit "#" instead of "Dial" when you're done 
dialing,
which I find to be much more intuitive than the "Dial"
softbutton.

Way to go cisco.  Will they be making this an industry 
standard next?

Even if Asterisk does the overlap stuff defined in RFC 
3578,
I seriously doubt you'll see the Cisco phones (or any 
hardware
phones, for that matter) doing it. The overlap stuff is 
really
designed for gateways from the PSTN, not end terminals.

SNOM phones have supported it for ages.


As a side note, I'll point out that the Pingtel phones 
let
you provision client-side digitmaps. Based on 
asterisk-like
pattern matching, you get to say how long a digit string
should be matched, and the phone will automatically dial
when it matches (no need to hit send!). You can even make
different patterns go different places, like:

972xxx : sip:[EMAIL PROTECTED]
214xxx : sip:[EMAIL PROTECTED]
489xxx : sip:[EMAIL PROTECTED]
1xx : sip:[EMAIL PROTECTED]
(To clarify: Dallas has three local area codes and 10 
digit
local dialing)

Still think overlap dialing is the go.  I wouldn't want to 
be responsible for administering 3000 phones with 
client-side digit maps, would you?

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP vs SCCP vs XML

[Jamie Carl]

On Mon, 25 Aug 2003 18:45:22 -0400
 "Ray Burkholder" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*

No, this is not the case currently with any of the Cisco 
SIP software 
loads that I am aware of.  If you find this to be 
incorrect, please 
let the list know.  Cisco has not deployed much of the 
featureset in 
their SCCP phones (such as paging/intercom) into the SIP 
phones due 
to lack of standards/interest/political capital.

JT


Ok, after further research in the 7960 administrators 
guide for SIP 5.1
(current is 5.3 and probably not changed much), they do 
state that
support is not provided for CiscoIPPhoneExecute in the 
current SIP load,
which is needed to make streaming channel 1 work. 
Bummer.

So, in looking around at HotDispatch.com, I see a number 
of companies
charging outrageous dollars for their own SCCP versions 
of a softphone.

Also, a while back, for $1000, a person could join 
Cisco's developer
program and gain access to SCCP docs.  Perhaps an 
Asterisk group member
has the funds available to attempt joining?  Then we 
could finish up on
some of the aborted attempts at SCCP integration, if the 
license
agreement allows this sort of development.

Perhaps, through a little creativity, it might be 
possible to use a SCCP
796x phone and not worry about SCCP.  With XML, screens 
could be
programmed to send responses back to *.  Then * could 
drive streaming
channel 1 directly and simulate the phone call.  So, on a 
SCCP phone,
you don't use SCCP, nor SIP.  You use XML.  Would that 
work?  Hopefully
soft button presses don't interfere with the streaming 
media.

Oh, and if it does work, then you can use multicasting to 
intercom a
number of phones simultaneously.

The thing I miss on SIP phones that was available on the 
Callmanager
version of 796x, is the ability to go off hook, dial some 
numbers, and
callmanager automatically dials the call.  The SIP 
version requires you
to go off hook, dial the digits, then press dial.  Any 
way around this
for 4, 7, 10 or 11 digit dialling?



Good question..   Does * support overlap dialing with SIP?

I have a feeling it does, I do vaguely remember getting an 
Address Incomplete response when not dialing enough 
digits.  I guess all you have to do is set your cisco 
phone for overlap dialing.  Hopefully there is an option 
for it in is config. 

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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RE: [Asterisk-Users] VIRUS ALERT

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*


*removes foot from mouth*

Appologies all round...

Yes, occasionally I do click reply to an old message, especially when
I'm using my remote webmail, only for the reason that I'm lazy and can't
be bothered typing in the mailing list address.  I don't see this a my
mail client groups by subject.

I still think he's a rude bastard tho and should learn some manners.

J

-Original Message-
From: Steve Meyers [mailto:[EMAIL PROTECTED]
Sent: Thursday, 21 August 2003 6:16 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VIRUS ALERT


*This message was transferred with a trial version of CommuniGate(tm)
Pro*
On Thu, 2003-08-21 at 02:01, Jamie Carl wrote:
> This IS a new thread 'bonehead'.

Actually, it wasn't.  He's correct in his complaint, however rude it may
have been.  It appears that you simply replied to a message and deleted
the subject and body, instead of starting a new message.  Email clients
that support threading get confused when you do this.

Here's a portion of your headers:

From: Jamie Carl <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
X-Mailer:  CommuniGate Pro WebUser Interface v.4.1
Message-ID:  <[EMAIL PROTECTED]>
In-Reply-To:  <[EMAIL PROTECTED]>
MIME-Version:  1.0
Content-Type:  text/plain; charset="ISO-8859-1"; format="flowed"
Content-Transfer-Encoding:  8bit
Subject: [Asterisk-Users] VIRUS ALERT

You see the "In-Reply-To:" header?  That says that your message is
actually a reply to the message with that Message-ID.  There's also a
References: header (not shown) that gives the Message-IDs of all
ancestors of your message.

Your mail client can use those headers to arrange the messages by
thread.  Steven obviously does this, and it annoys him when people break
his threading by not starting a new thread properly.

Steve
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Re: [Asterisk-Users] VIRUS ALERT

[Jamie Carl]

Oh, it's you again.  U know I really don't like you.  I 
just have to tell you that.  

This IS a new thread 'bonehead'.  This is MY virus alert 
asking whoever the dickhead that is sending me this crap 
to patch their pathetic system cause, like you do, it 
annoys me. (I bet it's you)

You might want to get your facts right before you start 
abusing people, or don't you have anything else to do with 
your pitiful existance?  You seem to spend more time 
abusing people about their posting ethics than actually 
helping people.  I think maybe YOUR ethics need to be 
revised.

Grow up!  And in future, if you don't have anything nice 
to say, don't say anything.

And btw, who uses the term 'luser' anymore?  I bet your 
first computer was an abacus, right?

J

On 20 Aug 2003 22:57:27 -0500
 Steven Critchfield <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Okay, bonehead, did you not see this thread started 2 
days ago? 

BTW, Have you not figured out how to start a new thread?

I can't believe so many lusers don't know how to start a 
thread.

Steven  

On Wed, 2003-08-20 at 18:29, Jamie Carl wrote:
Hi all,

I have recently received emails containing viruses in 
the 
last day or so.  As most of you should know there have 
been a slew of new viruses recently exploiting the 
Windoze 
OS.  Some are Windows XP specific but the one I received 
was the [EMAIL PROTECTED] worm.

Look at:
http://securityresponse.symantec.com/avcenter/venc/data/[EMAIL PROTECTED]
This worm is includes it's on SMTP transport and sends 
itself directly.  I have received this from the 
following 
addresses:

[EMAIL PROTECTED]
[EMAIL PROTECTED]
both originated from:

adsl-154-48-125.asm.bellsouth.net (68.154.48.125)

If this is your IP address and you have these addresses 
in 
your contact list, you are have a virus.

Tip for the future, do not open attachments from people 
you don't know!

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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--
Steven Critchfield <[EMAIL PROTECTED]>
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] VIRUS ALERT

[Jamie Carl]

And another one!

adsl-154-16-232.asm.bellsouth.net (68.154.16.232) 
   
If this is you, you ALSO have a virus!

Sorry to be sending these to the mailing list, but these 
appear to from the US and this is my only communication 
with the US, so logical would have it that it's someone on 
the mailing list.

Either that or one of the hundreds of people that send me 
ads for viagra or penis enlargements, but they never 
respond to my emails.

J

On Wed, 20 Aug 2003 20:32:52 -0400
 Jon Pounder <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Here's a better tip - don't use MS outhouse that likes to 
open attachments for you.

Oh yeah, that's a feature not a bug, I forgot.



At 08:29 PM 8/20/2003 -0400, you wrote:
On Wednesday 20 August 2003 07:29 pm, Jamie Carl wrote:

Tip for the future, do not open attachments from people
you don't know!
Regards,

Jamie Carl


Oh my god...I don't believe it.

Really...just people you don't know.

Don't open attachment's from people you know if there is 
no personal
information in the mail.

If there are obvious spelling errors or there is anything 
out of the ordinary
for the person you know.

Vague and strange comments that don't make any sense.

Common people.

Viruses etc. have been around for over 10-years.

This is old hat.
--
Emotions are alien to me.  I'm a scientist.
   -- Spock, "This Side of Paradise", 
stardate 3417.3

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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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[Asterisk-Users] VIRUS ALERT

[Jamie Carl]

Hi all,

I have recently received emails containing viruses in the 
last day or so.  As most of you should know there have 
been a slew of new viruses recently exploiting the Windoze 
OS.  Some are Windows XP specific but the one I received 
was the [EMAIL PROTECTED] worm.

Look at:
http://securityresponse.symantec.com/avcenter/venc/data/[EMAIL PROTECTED]
This worm is includes it's on SMTP transport and sends 
itself directly.  I have received this from the following 
addresses:

[EMAIL PROTECTED]
[EMAIL PROTECTED]
both originated from:

adsl-154-48-125.asm.bellsouth.net (68.154.48.125)

If this is your IP address and you have these addresses in 
your contact list, you are have a virus.

Tip for the future, do not open attachments from people 
you don't know!

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP QUESTION

[Jamie Carl]

In sip.conf:

canreinvite=no

And u're done.

J

On Tue, 19 Aug 2003 18:02:18 -0500 (CDT)
 Brian West <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Let me try this once again. :P  The reason I wanted 
everything to go thru
the * server is so you can monitor calls with 
res_monitor.

bkw

On Wed, 20 Aug 2003, Jamie Carl wrote:

Seeing as no one else has replied, I figured I may give 
it
a shot.  At least it'll start something.

Now, correct me if I'm wrong someone, but as far as I
understand in this situation you can do both.  Normally
the RTP packets would be swtiched through *, but you can
set in you sip.conf file the 'canreinvite=yes' option
which will allow the RTP stream to be direct if a
compatible codec is negotiated.
I'll double check if I ever get my server up and running
again.
J

On Tue, 19 Aug 2003 11:17:20 -0500
  "Jorge Cisneros Flores" <[EMAIL PROTECTED]> 
wrote:
>Hi
>
>
>   Is posible to make a call from site A to Site C, and
>my question is, the rtp data is from A to C or is from 
A
>to B to C
>
>
>
>
>Site A Site B
> Site C
> 
  ata186<>FW<->Asterisk<->FW<--->ata186
>
>Thanks

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP QUESTION

[Jamie Carl]

Seeing as no one else has replied, I figured I may give it 
a shot.  At least it'll start something.

Now, correct me if I'm wrong someone, but as far as I 
understand in this situation you can do both.  Normally 
the RTP packets would be swtiched through *, but you can 
set in you sip.conf file the 'canreinvite=yes' option 
which will allow the RTP stream to be direct if a 
compatible codec is negotiated.

I'll double check if I ever get my server up and running 
again.

J

On Tue, 19 Aug 2003 11:17:20 -0500
 "Jorge Cisneros Flores" <[EMAIL PROTECTED]> wrote:
Hi 

  Is posible to make a call from site A to Site C, and 
my question is, the rtp data is from A to C or is from A 
to B to C



   Site A Site B 
Site C
   ata186<>FW<->Asterisk<->FW<--->ata186

Thanks
Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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RE: [Asterisk-Users] SIP agent logging into queue?

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*

And what does your post have to do with SIP?

Where did u get H323 from?

Learn to read!

J (aka mailing list etiquette nazi)

-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Monday, 18 August 2003 4:51 PM
To: Asterisk List
Subject: Re: [Asterisk-Users] SIP agent logging into queue?


*This message was transferred with a trial version of CommuniGate(tm)
Pro*
On Mon, 2003-08-18 at 08:24, Sebastian Filzek wrote:
> Heya,
>
> I'm just playing with a SIP phone.  When I log into my queue from a
> SIP agent it appends some sort of data to the end of the SIP name.
> e.g. SIP/sablaptop-2ac0. I didn't add the '2ac0', asterisk did. When I
> log out of the queue, it uses a different ID (e.g. SIP/sablaptop-5207)
> and therefore does not log out.
>
> Does anyone know what the data tacked on the end of the SIP name is
> and how to stop it?
>
What does this have to do with H323?

Learn to post
--
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Recomendations for an ISDN-PBX to usewith asterisk

[Jamie Carl]

Bugga, it's definately a monday.  Replied to the wrong 
subject. (see below).

J

On Mon, 18 Aug 2003 09:42:05 +1000
 "Jamie Carl" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*

An 'ISDN' phone?  You mean a handset that actually 
support ISDN?  Didn't know they had these, and if they do 
I'm sure they wouldn't be cheap.

Are you talking BRI or PRI?  I'm guessing BRI which means 
you're right, there is no 'card' to go in an Asterisk box 
that will do this.

However, you might want to look at getting a 
SIP/MGCP/H323 Gateway that supports BRI-ISDN.  There are 
some out there and personally I'd got for SIP.  That way 
u can terminate/originate calls straight from the 
Asterisk box. 
Save a few headache with interop to the other inferrior 
PABX. :)

J

On Sun, 17 Aug 2003 22:12:30 +0200
 Miernik <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone 
Support wrote:
Hello, and thank you for registering at gnophone.com. 
Your login information is listed below:

Username: miernik
Password: ***
IAX Phone Number: 17002916107
Please login as soon as possible to 
http://x.linux-support.net/directory/ to complete the 
registration phase and activate your gnophone account.
Hi, 
I'd be grateful if any of recipiens of this message, 
could give me some clues on this problem, as googling the 
Internet didn't give me any clues.

I am a Debian GNU/Linux user and have just installed 
Gnophone from 
http://packages.debian.org/unstable/sound/gnophone.html

I have gnophone 0.2.4+cvs.20020624-4

I have registered at 
http://www.gnophone.com/directory/createAccount.php and 
received the IAX phone number above.

I did login, and enter my data in the gnophone program 
Preferences/Telephone Settings as follows:

Use Asterix

Server: iaxtel.com	Port: 5036
Context: iaxtel		Prefix: Username: miernik	Password: 
 (same as above)
Peer(optional):	miernik	Sercret(optional)  (same 
as above)

I am unable to call anywhere. If I try to call any 
1700xxx number, the program sends packets like this 
(output of tethereal):

20:02:31.6224 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2262 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2355 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2902 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.5429 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:34.4928 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036

but no packets ever come back. szrenica.ctnet.pl 
(212.126.24.133) is my computer. 
These are the only packets I see. 
No packets are blocked on any firewall between me and the 
Internet, my computer is reachable from the internet, you 
can ping it from the internet now to check. UDP is not 
blocked. 
I can even log over SSH to the only firewall between me 
and the internet (I'm the admin there) and I can see on 
the external interface that there are also no incoming 
UDP packets to/from port 5036.

Also in my logs I get messages like this:

Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Any help would be apprecieated. I'm leaving gnophone 
running, in case anyone would try to call me.

BTW: Duging subscription th gnophone/iaxtel service, I 
checked the box to be subcribed to gnophone users mailing 
list, but I didn't get any info how to post to that list.

regards, 
Jan Macek

--
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jabberid:[EMAIL PROTECTED]
__ ICQ: 4004001 ___/__ tel: +48608233394 __/ 
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Re: [Asterisk-Users] no incoming packets & Sound: Recordingoverrun

[Jamie Carl]

I'm not near my * box at the moment, so can't check this, 
but IAXTEL isn't down again, is it?  Can you ping 
iaxtel.com.

J

On Sun, 17 Aug 2003 22:12:30 +0200
 Miernik <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone 
Support wrote:
Hello, and thank you for registering at gnophone.com. 
Your login 
information is listed below:

 Username: miernik
 Password: ***
 IAX Phone Number: 17002916107
Please login as soon as possible to 
http://x.linux-support.net/directory/ to complete the 
registration 
phase and activate your gnophone account.
Hi, 

I'd be grateful if any of recipiens of this message, 
could give me 
some clues on this problem, as googling the Internet 
didn't give me 
any clues.

I am a Debian GNU/Linux user and have just installed 
Gnophone from 
http://packages.debian.org/unstable/sound/gnophone.html

I have gnophone 0.2.4+cvs.20020624-4

I have registered at 
http://www.gnophone.com/directory/createAccount.php and 
received the 
IAX phone number above.

I did login, and enter my data in the gnophone program 
Preferences/Telephone Settings as follows:

Use Asterix

Server: iaxtel.com	Port: 5036
Context: iaxtel		Prefix: 
Username: miernik	Password:  (same as above)
Peer(optional):	miernik	Sercret(optional)  (same 
as above)

I am unable to call anywhere. If I try to call any 
1700xxx number, 
the program sends packets like this (output of 
tethereal):

20:02:31.6224 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2262 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2355 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2902 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.5429 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:34.4928 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036

but no packets ever come back. 
szrenica.ctnet.pl (212.126.24.133) is my computer. 

These are the only packets I see. 

No packets are blocked on any firewall between me and the 
Internet, my 
computer is reachable from the internet, you can ping it 
from the 
internet now to check. UDP is not blocked. 

I can even log over SSH to the only firewall between me 
and the 
internet (I'm the admin there) and I can see on the 
external interface 
that there are also no incoming UDP packets to/from port 
5036.

Also in my logs I get messages like this:

Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Any help would be apprecieated. I'm leaving gnophone 
running, in case 
anyone would try to call me.

BTW: Duging subscription th gnophone/iaxtel service, I 
checked the box 
to be subcribed to gnophone users mailing list, but I 
didn't get any 
info how to post to that list.

regards, 

Jan Macek

--
Miernik  
jabberid:[EMAIL PROTECTED]
__ ICQ: 4004001 ___/__ tel: +48608233394 __/ 
mailto:[EMAIL PROTECTED]
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Regards,

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Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] no incoming packets & Sound: Recordingoverrun

[Jamie Carl]

An 'ISDN' phone?  You mean a handset that actually support 
ISDN?  Didn't know they had these, and if they do I'm sure 
they wouldn't be cheap.

Are you talking BRI or PRI?  I'm guessing BRI which means 
you're right, there is no 'card' to go in an Asterisk box 
that will do this.

However, you might want to look at getting a SIP/MGCP/H323 
Gateway that supports BRI-ISDN.  There are some out there 
and personally I'd got for SIP.  That way u can 
terminate/originate calls straight from the Asterisk box. 
Save a few headache with interop to the other inferrior 
PABX. :)

J

On Sun, 17 Aug 2003 22:12:30 +0200
 Miernik <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone 
Support wrote:
Hello, and thank you for registering at gnophone.com. 
Your login 
information is listed below:

 Username: miernik
 Password: ***
 IAX Phone Number: 17002916107
Please login as soon as possible to 
http://x.linux-support.net/directory/ to complete the 
registration 
phase and activate your gnophone account.
Hi, 

I'd be grateful if any of recipiens of this message, 
could give me 
some clues on this problem, as googling the Internet 
didn't give me 
any clues.

I am a Debian GNU/Linux user and have just installed 
Gnophone from 
http://packages.debian.org/unstable/sound/gnophone.html

I have gnophone 0.2.4+cvs.20020624-4

I have registered at 
http://www.gnophone.com/directory/createAccount.php and 
received the 
IAX phone number above.

I did login, and enter my data in the gnophone program 
Preferences/Telephone Settings as follows:

Use Asterix

Server: iaxtel.com	Port: 5036
Context: iaxtel		Prefix: 
Username: miernik	Password:  (same as above)
Peer(optional):	miernik	Sercret(optional)  (same 
as above)

I am unable to call anywhere. If I try to call any 
1700xxx number, 
the program sends packets like this (output of 
tethereal):

20:02:31.6224 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2262 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2355 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.2902 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:33.5429 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036
20:02:34.4928 szrenica.ctnet.pl -> 12.37.165.130 UDP 
Source port: 5036  Destination port: 5036

but no packets ever come back. 
szrenica.ctnet.pl (212.126.24.133) is my computer. 

These are the only packets I see. 

No packets are blocked on any firewall between me and the 
Internet, my 
computer is reachable from the internet, you can ping it 
from the 
internet now to check. UDP is not blocked. 

I can even log over SSH to the only firewall between me 
and the 
internet (I'm the admin there) and I can see on the 
external interface 
that there are also no incoming UDP packets to/from port 
5036.

Also in my logs I get messages like this:

Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun
Any help would be apprecieated. I'm leaving gnophone 
running, in case 
anyone would try to call me.

BTW: Duging subscription th gnophone/iaxtel service, I 
checked the box 
to be subcribed to gnophone users mailing list, but I 
didn't get any 
info how to post to that list.

regards, 

Jan Macek

--
Miernik  
jabberid:[EMAIL PROTECTED]
__ ICQ: 4004001 ___/__ tel: +48608233394 __/ 
mailto:[EMAIL PROTECTED]
Sing a declaration against US invasion in Iraq:
http://www.moveon.org/declaration/
___
Asterisk-Users mailing list
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Windows Messenger

[Jamie Carl]

Check out X-Lite.  http://www.xten.com

Only the older (pre 5.x) versions of Messenger support 
SIP.  If u can get a copy of v4.7 that will work fine. 
Instructions on setup can be easily found by doing a 
search of this mailing list. (look for the keywork 'MSN').

I do however suggest X-Lite, for the sole reason that it 
isn't written by Micro$oft.

J

On Sun, 10 Aug 2003 19:03:41 -0400 (EDT)
 [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Can anyone provide me with a step by step on how to set 
up Windows 
Messenger on a Windows XP Pro box as a SIP client with 
asterisk? I'm 
interested in doing various tests of my asterisk server 
from the Windows 
perspective of the world.  In the alternative if someone 
could provide 
information on another Windows based fully functional 
easy to configure 
iax or SIP client that would suffice as well.

Thanks in advance.
AJ
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Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
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Re: [Asterisk-Users] Open G.729A codec

[Jamie Carl]

I've been taking another approach to this codec/bandwidth 
problem.  Instead of trying to get more codecs into 
Asterisk (which is always hard due to licencing) I've been 
trying to get vendors to implement GSM in their products. 

SNOM do GSM.
D-Link gave me the good old, "we have plans to support.. 
blah blah.. "

Any others I can start harrassing?

I'm not giving up though.  But we have to remember to 
attack this problem from all angles.

J

On Tue, 12 Aug 2003 22:43:18 -0500
 "Matthew Hardeman" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Are the VoiceAge people generally unpleasant to work with 
and geniunely
uncaring, or do they just fail to respond?

Matt Hardeman
PaperSoft
- Original Message - 
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 12, 2003 10:16 PM
Subject: Re: [Asterisk-Users] Open G.729A codec


> I made a mistake of buying it so that I can have a 
low-bandwidth
> well-tested codec for use on an IAX2 link. Then I've 
caused Digium lots
> of unwanted trouble, because hair stood on the back of 
my neck after
> reading the licensing agreement and seeing the .so 
library. Let's hope
> it gets better in the future!

Believe it or not, we worked hard to get that license 
agreement
*improved*.  I wish they took our concerns (and those of 
our customers)
more seriously.

Mark

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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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[Asterisk-Users] SIP Transfer

[Jamie Carl]

Ok, just been thinking about this and thought I would ask 
before trying it out again.

What is the state of SIP transfers?  By this I mean 
transfers initiated via SIP messages, not via DTMF and 
'#'.  

Last time I tried, on X-Lite, clicking the transfer button 
dropped the call.

Also, are/will both REFER and BYE/also methods be 
supported?  To me, the SIP way of transfering is alot 
nicer and it seems silly to me to have a transfer button 
on your SIP phone that u can't use.

Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] CODEC & DTMF

[Jamie Carl]

Hopefully these are all correct:

-No, a SIP phones cannot connect to Asterisk voicemail 
using G.729 if you do not have a licence.  You will need a 
licence for at least one channel to do this.
-DTMF is not related to the codec itself or how it works, 
however, inband DTMF is not recommended on compressed 
channels due to the possibility of distortion.  Outband is 
the way to go as the tones are capture and then 
regenerated locally.

Inband signaling is achieved thought the telephony voice 
channel, while outband is through some other type of 
communication channel. In the case of SIP it is the INFO 
method.

Hope this helps,

J

On Thu, 14 Aug 2003 09:03:15 -0700
 "George Lin" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*

Dear all,

I like to know if the DTMF option is related to the codec 
or not. Can a SIP
phone with g729 codec to access asterisk voicemail2 in 
case the asterisk
does not have g729 license ?? If yes, what is the DTMF 
option inband or
outband ??? Is there any successful experience ???

Regards,

George Lin

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Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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RE: [Asterisk-Users] How to Asterisk

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*
Ooh, can i answer this one?  Please??


RTFM! :)

http://www.digium.com/handbook-draft.pdf

hehe...

Regards,

Jamie Carl
Jazz-Inc.
Email:  [EMAIL PROTECTED]
Phone:  +61 414 365 466
Jabber: [EMAIL PROTECTED]




-Original Message-
From: prakashmodak_74 [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 12 August 2003 5:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] How to Asterisk


*This message was transferred with a trial version of CommuniGate(tm)
Pro*
Hello,

 I'm new user of asterisk. Can anybody pls tell me how to use asterisk
or any detail how to link

 i installed Asterisk-0.4.0 on i810 onboard sound card with Redhat 7.1.
when i type "asterisk -vvvc"  i get  *CLI> prompt




Prakash
Get Your Private, Free E-mail from Indiatimes at
http://email.indiatimes.com

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Before. Just log on to http://airsahara.indiatimes.com and Bid Now!

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Re: [Asterisk-Users] usrobotics modem and pstn

[Jamie Carl]

Rewrite the modem driver so it support full-duplex audio, 
because currently it doesn't.  :)

Basically, to get a modem working u need to setup 
/etc/asterisk/modem.conf.  It's all pretty self 
explanitory given the comments therein.  

But like I sed, even if you get it working, you'll only 
get one-way voice because it's not full-duplex.  (Unless 
this has been fixed since I tried it last).

Get an X100p.  They're cheap and work like 1000 times 
better than a modem.

J

On 12 Aug 2003 17:15:25 -0500
 santiago <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
hi,

i have a external usrobotics modem, i want to use it with 
asterisk to
interact with the pstn, 

what i have to do?

thanks,

--
santiago josé ruano rincón
administración servidores y servicios de internet
red de datos
universidad del cauca
 
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Re: [Asterisk-Users] Get faxed you faxing faxer!

[Jamie Carl]

Sorry to say this Gary, but I think you're missing the 
whole point behind Asterisk.  Why merely try and 
'recreate' what PABXs already do and have been doing for 
the last quadzillion years.  Asterisk, BY DESIGN, is 
trying to do, not only the same stuff, but MORE and 
BETTER.  

So, keeping this in mind, why is it so outrageously 
rediculous to ask?  If asterisk is MORE than just a PABX, 
won't that mean it's MORE appealing to users?  

I don't expect anything when it comes to Asterisk.  If a 
feature works, great, I have a new toy.  If it doesn't, I 
try and help fix it.  If it's not there, i 'suggest' it. 
It's all part of the open source process.

Oh, and as for PABXs in australia that support fax, well, 
Cisco have a nice little full IMS system that does it. 
But who wants to pay $1,000,000 for a PABX, right?

Jazz

On Tue, 12 Aug 2003 10:17:35 +1000
 "Gary" <[EMAIL PROTECTED]> wrote:
I always have a chuckle when I see this.

it probably could if someone sorts it out, but its 
reqally starting to
expect a lot.

I really cant think of any pab system (at least here in 
australia)
which has a builtin fax.  fax/phones sure, fax on 
computers sure, but
pab/fax m, I can just see my p100 sitting at home 
trying to deal
with faxes and phone calls.  (shudder..)

On Tue, 12 Aug 2003 10:23:05 +1000, Jamie Carl wrote:

All this talk of faxing has started me thinking (this is 
always a bad thing) and I've come up with a question.

Now, I know Asterisk can detect and route faxes, to a 
'fax' extension and all that.  But can Asterisk be used 
to 
'receive' faxes?  

I know there was some talk about this just over a year 
ago 
and I'm wondering if anything came of it.  It would be 
nice to be able to receive faxes with asterisk and then 
have them filer through and AGI script and emailed or 
dumped in a directory/database/whatever.

I mean for me, currently, I have a fax/modem hooked up to 
the same machine asterisk is on, and it get's sent faxes 
via my FXS port.  It would be nice to be able to pull off 
the modem and throw it away. (or auction it on ebay)
 
Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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Re: [Asterisk-Users] SNOM200 firmware roll back!!

[Jamie Carl]

Why not ask them??

Christian Stredicke is the man to talk to.

Although, i just had a look at the SNOM website and 1.16w 
is still there to download.

J

On Sun, 10 Aug 2003 16:59:37 +
 "WipeOut ." <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Look like SNOM have rolled back the firmware version of 
the 200's from 1.16w to 1.16q..

Anyone know why?
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Re: [Asterisk-Users] Get faxed you faxing faxer!

[Jamie Carl]

I've got an old P200Pro and a Celeron 333 CPU and MB 
sitting around if you want them. :)

Jazz

On Tue, 12 Aug 2003 11:31:37 +1000
 "Gary" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
sorry jamie, i didn;t miss the point, i just didn't put 
in the
smiley...

you might have missed what one of asterisk machines is 
actually running
on...

p100 was refering to a pentium 100 processor, doing other 
stuff as well
asterisk... throw a fax at it and would probably limp to 
computer
heaven !!  (even running linux) ... it runs normally at 
75% cpu
utilisation   :-)

On Tue, 12 Aug 2003 10:52:46 +1000, Jamie Carl wrote:

Sorry to say this Gary, but I think you're missing the 
whole point behind Asterisk.  Why merely try and 
'recreate' what PABXs already do and have been doing for 
the last quadzillion years.  Asterisk, BY DESIGN, is 
trying to do, not only the same stuff, but MORE and 
BETTER.  

So, keeping this in mind, why is it so outrageously 
rediculous to ask?  If asterisk is MORE than just a PABX, 
won't that mean it's MORE appealing to users?  

I don't expect anything when it comes to Asterisk.  If a 
feature works, great, I have a new toy.  If it doesn't, I 
try and help fix it.  If it's not there, i 'suggest' it. 
It's all part of the open source process.

Oh, and as for PABXs in australia that support fax, well, 
Cisco have a nice little full IMS system that does it. 
But who wants to pay $1,000,000 for a PABX, right?

Jazz

On Tue, 12 Aug 2003 10:17:35 +1000
 "Gary" <[EMAIL PROTECTED]> wrote:
I always have a chuckle when I see this.

it probably could if someone sorts it out, but its 
reqally starting to
expect a lot.

I really cant think of any pab system (at least here in 
australia)
which has a builtin fax.  fax/phones sure, fax on 
computers sure, but
pab/fax m, I can just see my p100 sitting at home 
trying to deal
with faxes and phone calls.  (shudder..)

On Tue, 12 Aug 2003 10:23:05 +1000, Jamie Carl wrote:

All this talk of faxing has started me thinking (this is 
always a bad thing) and I've come up with a question.

Now, I know Asterisk can detect and route faxes, to a 
'fax' extension and all that.  But can Asterisk be used 
to 
'receive' faxes?  

I know there was some talk about this just over a year 
ago 
and I'm wondering if anything came of it.  It would be 
nice to be able to receive faxes with asterisk and then 
have them filer through and AGI script and emailed or 
dumped in a directory/database/whatever.

I mean for me, currently, I have a fax/modem hooked up to 
the same machine asterisk is on, and it get's sent faxes 
via my FXS port.  It would be nice to be able to pull off 
the modem and throw it away. (or auction it on ebay)
 
Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
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Re: [Asterisk-Users] X-Lite <-> Snom200

[Jamie Carl]

Yes, over a LAN.  It does it with both g.711 and GSM which 
both used to work.  Havn't had a chance to have a REAL 
good look into it though.

J

On Wed, 06 Aug 2003 14:33:47 +
 "WipeOut ." <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Dunno what I'm doing wrong here but I just did an 
upgrade to the latest
version and now I get no audio at all!
I havn't changed a single thing.  Is there anything 
special I need to do
to get this to work again?

I get a quick 'chirp' of audio, which you can tell is 
what I'm
connecting to, (ie MOH), but then nothing.

Regards,

Jamie Carl
Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
Phone:  +61 414 365 466
Jabber: [EMAIL PROTECTED]
Are you connecting to * over a LAN?? I have experienced 
the "chirp" when the phone was trying to use G.711 over a 
dial up link so there was not enough bandwidth..

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Re: [Asterisk-Users] SNOM200 firmware roll back!!

[Jamie Carl]

Why not ask them??

Christian Stredicke is the man to talk to.

Although, i just had a look at the SNOM website and 1.16w 
is still there to download.

J

On Sun, 10 Aug 2003 16:59:37 +
 "WipeOut ." <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Look like SNOM have rolled back the firmware version of 
the 200's from 1.16w to 1.16q..

Anyone know why?
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[Asterisk-Users] Get faxed you faxing faxer!

[Jamie Carl]

All this talk of faxing has started me thinking (this is 
always a bad thing) and I've come up with a question.

Now, I know Asterisk can detect and route faxes, to a 
'fax' extension and all that.  But can Asterisk be used to 
'receive' faxes?  

I know there was some talk about this just over a year ago 
and I'm wondering if anything came of it.  It would be 
nice to be able to receive faxes with asterisk and then 
have them filer through and AGI script and emailed or 
dumped in a directory/database/whatever.

I mean for me, currently, I have a fax/modem hooked up to 
the same machine asterisk is on, and it get's sent faxes 
via my FXS port.  It would be nice to be able to pull off 
the modem and throw it away. (or auction it on ebay)
 
Regards,

Jamie Carl
Jazz Inc.
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RE: [Asterisk-Users] X-Lite <-> Snom200

[Jamie Carl]

*This message was transferred with a trial version of CommuniGate(tm) Pro*
Dunno what I'm doing wrong here but I just did an upgrade to the latest
version and now I get no audio at all!
I havn't changed a single thing.  Is there anything special I need to do
to get this to work again?

I get a quick 'chirp' of audio, which you can tell is what I'm
connecting to, (ie MOH), but then nothing.


Regards,

Jamie Carl
Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
Phone:  +61 414 365 466
Jabber: [EMAIL PROTECTED]

-Original Message-
From: WipeOut . [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 6 August 2003 10:05 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] X-Lite <-> Snom200


*This message was transferred with a trial version of CommuniGate(tm)
Pro*
> Quoting WipeOut:
> > Hi,
> >
> > I have just been playing with the latest X-Lite.. It works fine
> > with Asterisk..
> >
> > As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only
> > one that didn't work.. not sure why..
>
> Did you get Speex working? I've tried, but although I can get it to
connect,
> there is no audio :(
>
> Jamie
>

No I didn't try Speex, but the result you got is the same as what I got
when trying iLBC..

I was glad to find that GSM now has a much better quality than when I
last tried X-Lite many months ago..

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[Asterisk-Users] IAXTEL testing

[Jamie Carl]

Hi
all,
 
Just a quick
one.  Should I be able to call myself through IAXTEL using my 1700
number?  I'm behind a NAT firewall and can call other numbers, I just want
to test incoming calls somehow to make sure I can accept them from
IAXTEL.
 
Regards,Jamie Carl

  
  
  
  
  

  Email:

  [EMAIL PROTECTED]
  

  Phone:

  +61 414 365 466
  

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  [EMAIL PROTECTED]

 

Re: [Asterisk-Users] ANI matching trouble

[Jamie Carl]

Shouldn't the priority also be different for each entry?

This would make it:

exten => 4044633/_619.,1,OurApp,sandiego-queue
exten => 4044633/_858.,2,OurApp,sandiego-queue
exten => 4044633/_213.,3,OurApp,losangeles-queue
exten => 4044633,4,OurApp,default-queue
This should work I would think.  Give it a shot, if it 
doesn't remove the _'s from the ANI pattern.

Jamie

On Wed, 28 May 2003 22:39:19 -0500 (CDT)
 Mark Spencer <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
exten => 4044633/_619.,1,OurApp,sandiego-queue
exten => 4044633/_858.,1,OurApp,sandiego-queue
exten => 4044633/_213.,1,OurApp,losangeles-queue
exten => 4044633/_.,1,OurApp,default-queue
but it didn't seem to work.  Every call went to the 
default queue.
Take out the _. rule and just leave it 4044633 and it 
should work fine.
Not postive the _ is required on matching the callerid 
part, but honestly
i just don't remember.

Mark

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Re: [Asterisk-Users] REMOVE

[Jamie Carl]

No!  You're outnumbered and trapped!

On Wed, 28 May 2003 10:22:33 -0700 (PDT)
 Patrick Tabor <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*

--- Gary <[EMAIL PROTECTED]> wrote:
I just had a thought, but haven't tried it yet

we are able to include one conf file into another...

this leads to the thought can we use the same file
to include into
iax.conf as well as sif.conf ?
does the various differences actually have an effect
of the application
reading the file ?
or another way

will nonexisitant controlcommands have an adverse
effect
eg: in SIP.conf having nat=yes, what happens if that
was in the same
user stuff in iax.conf ??
Gary
.


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Re: [Asterisk-Users] The Phantom Call.. T1 card too

[Jamie Carl]

I take back my previous comments then, Steven.  I really 
need thread concatination on my webmail system. :)

Jamie

On 27 May 2003 16:50:19 -0500
 Steven Critchfield <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
Sorry, I seem to have responded to the wrong thread 
there. Should have
been part of the Duplicate numbers thread.

On Tue, 2003-05-27 at 12:36, Joe Antkowiak wrote:
No popping/bad audio on this one, clear as can be, 
asterisk just decides to
pick up the channel after about a minute and use the "s" 
extension in the
context...  immediate=no is set on this channel.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf 
Of Steven
Critchfield
Sent: Tuesday, May 27, 2003 1:36 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] The Phantom Call.. T1 card 
too

I had a similar problem when there was timing problems 
with my T100P.
You could also hear lots of popping and generally bad 
audio during the
dial tone. After we fixed the timing problem, the audio 
was clear as
could be, and the problems went away. Of course this 
doesn't fix a X100P
as it is strictly analog.

On Tue, 2003-05-27 at 12:03, Joe Antkowiak wrote:
> I've had the same thing happen, only on the single 
port T1 card and a
> channel bank, and one of the FXO channels also having 
a phone attached
> elsewhere...
> 
> I just wound up putting that channel in a different 
context and running
> 
> Exten => s,1,Hangup
> 
> (I'm just using the line for outbound dialing)
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On 
Behalf Of Tamas Levente
> Sent: Tuesday, May 27, 2003 11:45 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] The Phantom Call..
> 
> Same thing happened with me too. X100P. Same US tones
> Sometimes it gets into the voicemail too:)) And the 
voicemail record 3
> minutes tone, after 1.5minutes it's service not 
available or something
> similar.
> Is there a fix for that?
> 
> - Original Message - 
> From: "Mark Street" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, May 27, 2003 5:39 PM
> Subject: Re: [Asterisk-Users] The Phantom Call..
> 
> 
> > Funny,  I just noticed this happening on my box with 
2 X101P's installed
> and a
> > phone connected to the same line as one of the 
X101P's.  I pick up the
> phone
> > after 1 ring, or call someone.  After a minute or 
two * picks up the
line
> and
> > starts the greeting.  I pull the plug on the 
asterisk box to
continue
> the
> > conversation.  I just noticed it happening a couple 
of weeks ago.  US
> > dialtone here...
> >
> > On Tuesday 27 May 2003 08:13, Mark Spencer wrote:
> > > > Could it be that the X100P is detecting the UK 
dial tone as a ring??
> > > > or Has anyone else had a similar problem when 
using the X100P/S100U
> > > > combination??
> > >
> > > It's possible there is *something* on the line 
that is confusing
> Asterisk
> > > into thinking a ring takes place.  You might try 
adjusting the value
of
> > > PEGCOUNT in wcfxo.c to a higher value (say, 10).
> >
> > -- 
> > Mark Street, D.C.
> > Red Hat Certified Engineer
> > Cert# 807302251406074
> > --
> > Key fingerprint = 3949 39E4 6317 7C3C 023E  2B1F 
6FB3 06E7 D109 56C0
> > GPG key http://www.streetchiro.com/pubkey.asc
> >
> > ___
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> > 
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> >
> 
> 
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Re: [Asterisk-Users] The Phantom Call.. T1 card too

[Jamie Carl]

Then why bring it up?

What has any of this to do with the original topic?

Jamie

On 27 May 2003 12:35:57 -0500
 Steven Critchfield <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
I had a similar problem when there was timing problems 
with my T100P.
You could also hear lots of popping and generally bad 
audio during the
dial tone. After we fixed the timing problem, the audio 
was clear as
could be, and the problems went away. Of course this 
doesn't fix a X100P
as it is strictly analog.

On Tue, 2003-05-27 at 12:03, Joe Antkowiak wrote:
I've had the same thing happen, only on the single port 
T1 card and a
channel bank, and one of the FXO channels also having a 
phone attached
elsewhere...

I just wound up putting that channel in a different 
context and running

Exten => s,1,Hangup

(I'm just using the line for outbound dialing)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf 
Of Tamas Levente
Sent: Tuesday, May 27, 2003 11:45 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] The Phantom Call..

Same thing happened with me too. X100P. Same US tones
Sometimes it gets into the voicemail too:)) And the 
voicemail record 3
minutes tone, after 1.5minutes it's service not 
available or something
similar.
Is there a fix for that?

- Original Message - 
From: "Mark Street" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 27, 2003 5:39 PM
Subject: Re: [Asterisk-Users] The Phantom Call..

> Funny,  I just noticed this happening on my box with 2 
X101P's installed
and a
> phone connected to the same line as one of the 
X101P's.  I pick up the
phone
> after 1 ring, or call someone.  After a minute or two 
* picks up the line
and
> starts the greeting.  I pull the plug on the 
asterisk box to continue
the
> conversation.  I just noticed it happening a couple of 
weeks ago.  US
> dialtone here...
>
> On Tuesday 27 May 2003 08:13, Mark Spencer wrote:
> > > Could it be that the X100P is detecting the UK 
dial tone as a ring??
> > > or Has anyone else had a similar problem when 
using the X100P/S100U
> > > combination??
> >
> > It's possible there is *something* on the line that 
is confusing
Asterisk
> > into thinking a ring takes place.  You might try 
adjusting the value of
> > PEGCOUNT in wcfxo.c to a higher value (say, 10).
>
> -- 
> Mark Street, D.C.
> Red Hat Certified Engineer
> Cert# 807302251406074
> --
> Key fingerprint = 3949 39E4 6317 7C3C 023E  2B1F 6FB3 
06E7 D109 56C0
> GPG key http://www.streetchiro.com/pubkey.asc
>
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Re: [Asterisk-Users] VOIP phone suppliers in the UK?

[Jamie Carl]

If u're itching to get started testing SIP, prolly best to 
go for a softphone until you can get u're hands on a 
hardware version.  Least that way you can get your head 
around how to setup Asterisk to use SIP correctly.

I use X-Ten X-LITE which is free and it seems to work 
rather well.  You can get it from http://www.xten.com

Jamie

On Wed, 28 May 2003 12:02:11 +0100
 "nathan" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*



http://www.solwise.co.uk/voip_phones.htm

I can recommend the supplier (used them for ADSL 
routers)
but can't say anything about the specific product 
listed.

Especially as it isn't available for a few more 
weeks.

They are generally very savvy about their products 
though, being a
small company
Mike

I was looking at Solwise, and am very interested in their 
IP2005
SIP phone for £99+vat. They delayed the release from 
mid-may to
mid-june of the IP2005. I guess I can wait, but I'm eager 
to start
testing a SIP phone with Asterisk :-)

I've never used them before but they do have a good 
reputation, and 
as you've mentioned they know what they're talking about.

-Nathan

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Re: [Asterisk-Users] Gastman windows build?

[Jamie Carl]

This build is about 200 years old.  I've tried it and it 
actually doesn't work for me.  CVS builds work fine 
though.

Jamie

On Wed, 28 May 2003 00:40:44 +1000
 "Shaun Ewing" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*

- Original Message - 
From: "Dave Packham" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, May 28, 2003 12:03 AM
Subject: [Asterisk-Users] Gastman windows build?


is the win32 binary on * website suilt from the latest 
gastman code?

Thanks

Dave
This is the latest prebuilt binary I could find:

ftp://ftp.digium.com/pub/gastman/gastman-win32-0.2.1.zip

--Shaun

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Re: [Asterisk-Users] ANI matching trouble

[Jamie Carl]

I was just thinking that.  Shouldn't this be a feature? 
I'm sure coding it would be a cut and past job. :)

Another one for the TO-DO list Mark. 
:)

Jamie

On Wed, 28 May 2003 16:45:44 -0700
 John Todd <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of 
CommuniGate(tm) Pro*
I don't think you can use wildcards in the ANI matching 
areas, though I'd be happy if this were the case.

You'll probably need to write an AGI that hands back an 
appropriate variable set to something that a Goto can 
parse.

The use of wildcards in ANI matches would be darn handy, 
though.

JT

Hi.  I need to send calls to different programs depending 
on where the
call originates.  For example, I need calls from San 
Diego (NPA 619 and
858) to to be routed differently than L.A. calls.  I 
tried entries like:

exten => 4044633/_619.,1,OurApp,sandiego-queue
exten => 4044633/_858.,1,OurApp,sandiego-queue
exten => 4044633/_213.,1,OurApp,losangeles-queue
exten => 4044633/_.,1,OurApp,default-queue
but it didn't seem to work.  Every call went to the 
default queue.

I also tried
exten => 4044633/_619XXX,1,OurApp,sandiego-queue
to no avail.
It did work if I put a specific number in there:
exten => 4044633/6193644788,1,OurApp,sandiego-queue
but of course I can't list every possible number.

What am I doing wrong?  Thanks...
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Regards,

Jamie Carl
Jazz Inc.
Email:  [EMAIL PROTECTED]
Web:www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: [EMAIL PROTECTED]
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[Asterisk-Users] Full-Duplex on Voice Modem

[Jamie Carl]

Hey all,

Just a quick question.

I've managed to configure my generic rockwell voice modem with asterisk and
can make calls out to the PSTN through it.  Thing is I get one way voice.
Is this a problem with the voice modem not being capable of full-duplex
voice, or is it the voice modem channel driver for asterisk?

Anyone know what is required to get this working as being in Australia, it's
hard to get cheap analog hardware to link to the PSTN and being able to
stick a couple of voice modems in a box would be a nice way of doing it.

Regards,

Jamie Carl
Email:  [EMAIL PROTECTED]
PH: +61-414-365-466

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[Asterisk-Users] PSTN Card Suggestions

[Jamie Carl]

Hey all,

I'm now trying to source some PCI cards that I can use to terminate into the
PSTN.

I know about digiums Wildcard range, but the X100P has only a single FXO
port.  Is there anything out there I can use that has say, 4 ports?  Or am I
going to have to use 4 cards?  Also, where can I get them in Australia and
how much?

Also, what are people using for ISDN BRI interfaces?

thanks peeps,

Regards,

Jamie Carl
Email:  [EMAIL PROTECTED]
PH: +61-414-365-466

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RE: [Asterisk-Users] MSN Messenger Versions

[Jamie Carl]

Thanx heaps everyone!!

--Jamie

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Wade Weppler
> Sent: Friday, 7 March 2003 1:24 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] MSN Messenger Versions
>
>
> I'll try to be a little more specific:
>
> http://www.microsoft.com/exchange/downloads/2000/IMClient47.asp
>
> This one works for me.
>
> -wade
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk
> > Sent: Thursday, March 06, 2003 8:45 AM
> > To: [EMAIL PROTECTED]; Jamie Carl
> > Subject: Re: [Asterisk-Users] MSN Messenger Versions
> >
> > microsoft.com
> >
> > On Thursday 06 March 2003 14:20, Jamie Carl wrote:
> > > I seem to have misplaced my copy.  Isn't that always the way.
> > >
> > > Anyone know where I can get an older 4.x version from?
> > >
> > > Regards,
> > >
> > > Jamie Carl
> > > Email:[EMAIL PROTECTED]
> > > PH:   +61-414-365-466
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:[EMAIL PROTECTED] Behalf Of William X
> > > > Walsh
> > > > Sent: Thursday, 6 March 2003 11:50 PM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: Re: [Asterisk-Users] MSN Messenger Versions
> > > >
> > > >
> > > >
> > > > 5 has no SIP support, go back to 4.6 or 4.7
> > > >
> > > > On Thu, 2003-03-06 at 04:20, Jamie Carl wrote:
> > > > > Hey all,
> > > > >
> > > > > Just wanted to know what versions of MSN Messenger people have
> > > >
> > > > working with
> > > >
> > > > > Asterisk.
> > > > >
> > > > > I had 4.5 or so working but with the new 5.whatever it seems to
> > > >
> > > > be a pain in
> > > >
> > > > > the ass.
> > > > >
> > > > > Thanks..
> > > > >
> > > > > Regards,
> > > > >
> > > > > Jamie Carl
> > > > > Email:[EMAIL PROTECTED]
> > > > > PH:   +61-414-365-466
> > > > >
> > > > > ___
> > > > > Asterisk-Users mailing list
> > > > > [EMAIL PROTECTED]
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > --
> > > > William Walsh <[EMAIL PROTECTED]>
> > > > Jabber: [EMAIL PROTECTED]
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > --
> > Roy Sigurd Karlsbakk, Datavaktmester
> > ProntoTV AS - http://www.pronto.tv/
> > Tel: +47 9801 3356
> >
> > Computers are like air conditioners.
> > They stop working when you open Windows.
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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>

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RE: [Asterisk-Users] MSN Messenger Versions

[Jamie Carl]

I seem to have misplaced my copy.  Isn't that always the way.

Anyone know where I can get an older 4.x version from?

Regards,

Jamie Carl
Email:  [EMAIL PROTECTED]
PH: +61-414-365-466

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of William X
> Walsh
> Sent: Thursday, 6 March 2003 11:50 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] MSN Messenger Versions
> 
> 
> 
> 5 has no SIP support, go back to 4.6 or 4.7
> 
> On Thu, 2003-03-06 at 04:20, Jamie Carl wrote:
> > Hey all,
> > 
> > Just wanted to know what versions of MSN Messenger people have 
> working with
> > Asterisk.
> > 
> > I had 4.5 or so working but with the new 5.whatever it seems to 
> be a pain in
> > the ass.
> > 
> > Thanks..
> > 
> > Regards,
> > 
> > Jamie Carl
> > Email:  [EMAIL PROTECTED]
> > PH: +61-414-365-466
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> William Walsh <[EMAIL PROTECTED]>
> Jabber: [EMAIL PROTECTED]
> 
> ___
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[Asterisk-Users] MSN Messenger Versions

[Jamie Carl]

Hey all,

Just wanted to know what versions of MSN Messenger people have working with
Asterisk.

I had 4.5 or so working but with the new 5.whatever it seems to be a pain in
the ass.

Thanks..

Regards,

Jamie Carl
Email:  [EMAIL PROTECTED]
PH: +61-414-365-466

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[Asterisk-Users] Not so important info

[Jamie Carl]

Hey all,

Quite a few people have been contacting me off-list and by phone regarding
my work with the SNOM phones and Asterisk, here in Australia.  Unfortunately
I no longer work for Salient Networks due to market instability and cannot
be reached at my usual email address and phone number.

If anyone wishes to contact me for further assistance, (or to offer jobs,
hehe) I can be reached at this email address.


Regards,

Jamie Carl
Email:  [EMAIL PROTECTED]
PH: +61-414-365-466

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RE: [Asterisk-Users] Snom 200

[Jamie Carl]

I take it this is in the USA?

If you want to get them from Australia, my company sells them here.

haha!
How's that for some shameless pimping!?

Jamie Carl
Network Engineer
Salient Networks Pty Ltd
Level 2, 4-10 Bridge Street
Pymble, Australia
NSW 2073
Support:  1800 648 287
Phone:+61 2 9144 2622
Mobile:   +61 414 365 466
Fax:  +61 2 9144 7266


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jerry Gibson
Sent: Thursday, 27 February 2003 10:54 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Snom 200


David:

I talked to one of the key people at ABP a few minutes ago. They had a
snow storm in Dallas today, and everything is closed  down. If you send
an e-mail to   [EMAIL PROTECTED],  they are working from home, and will
respond. They will be in the office tomorrow. 

Thanks,
Jerry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Davis
Sent: Wednesday, February 26, 2003 6:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Snom 200


Does anyone know who sells the Snom 200 (or a better suggestion) other
than ABP(http://www.abpintl.com/)?  They aren't answering their phones
and email today and I need to get 5 of these sent out very soon.

David


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