access to certain configuration files.
On Wed, Sep 28, 2016 at 01:24:58PM -0400, Ahmed Munir wrote:
> Hi Andrew and Willy,
>
> Thanks for sharing the info.
>
> As for enabling radius server debugging 'radiusd -X', made some test calls
> don't see the radiuscli
adius.c:208 radius_log: Unable to create RADIUS record. CDR
> > not recorded!
> >
> > Please advise if I missed out anything.
> >
> >
> > Date: Mon, 26 Sep 2016 12:09:34 +0200
> >> From: Willy Offermans
> >> To: Asterisk Users Mailing List - Non-Commercia
Hello Ahmed,
On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote:
> Hi,
>
> I've recently setup Asterisk with Radius CDR by following the document:
> https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend.
>
> The issue currently I'm facing is after turning on the debug getting
>
Dear D'Arcy J.M. Cain and asterisk friends,
On Tue, Aug 30, 2016 at 09:56:05AM -0400, D'Arcy J.M. Cain wrote:
> I have an extension that looks like this:
>
> exten => 55,1,Verbose(Door buzzer calling)
> same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
>
> The idea is that any of the three
Hello Kevin, hello asterisk friends,
On Sat, Jun 11, 2016 at 05:33:54AM +, Kevin Long wrote:
>
>
> Is it possible to configure Asterisk such that numerical extensions and/or
> usernames, would be populated from LDAP, as well as authenticate the
> endpoints where the “SIP secret” is equa
Dear asterisk friends,
Can someone tell me whether asterisk supports PAM authentication or not?
--
Met vriendelijke groeten,
With kind regards,
Mit freundlichen Gruessen,
Will
*
W.K. Offermans
Powered by
Dear asterisk friends,
I like to use asterisk and to do authentication, authorization and
accounting (AAA) for it with freeradius. I looked for any documentation on
the net, but could not find much useful and detailed information.
I have made a first shot with radiusclient-ng. I configured cdr
I can not see the callerID being passed.
Any ideas?
WW
--
Willy Wouters, PhD
Asterisk Telephony
Web Applications
MAGU ENTERPRISES
Tel: 713-474-1534
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--Bandwidth and Coloc
7;m looking for something to chomp this automatically.
--
Willy Wouters, PhD
Asterisk Telephony
Web Applications
MAGU ENTERPRISES
Tel: 713 474-1534
Fax: 501 665-1544
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talk ... He's just there to follow up on
the appointment and 'qualify' the customer to see if we are
worthy of their cheap service. After I looked at their
website, I can hear 'quack quack'.
Cheers,
WW
Willy Wouters
ypOne Publishing
_
ng to do that. Iconnecthere just gives
us (USA based) international dialtone.
Willy
- Original Message Follows -
> I've been to the WIKI and I've searched the archives.
>
> Is anyone on the list successfully using iconnecthere
> behind NAT?
>
> I was, for over a y
ded 'r' to all my Dial commands.
Thanks again,
Willy
> What does your dial extension look like?
>
> best regards
>
> kapejod
> --
> Klaus-Peter Junghanns
>
> CEO, CTO
> Junghanns.NET GmbH
> Breite Strasse 13a - 12167 Berlin - Germany
> fon: (de) +49 3
ation' which
presumably is the recording I am never hearing.
> Any ideas anyone?
> It kinda annoys our users, since they like to *know* when
> they dial an invalid number.
> TIA,
> WW
>
> Willy Wouters
> ypOne Publishing
>
> ___
gh the T1 / PRI interface however, I
continually hear ringing, and then the call gets hungup.
Any ideas anyone?
It kinda annoys our users, since they like to *know* when
they dial an invalid number.
TIA,
WW
Willy Wouters
ypOne Publishing
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Asterisk-Use
,2,Dial(SIP/user1|60|r)
But the empty time works as well. It will just ring
forever.
Cheers,
Willy
- Original Message Follows -
>
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
>
> > Explicitly answer the line. If that doesn't handle
> > inband audio,
> Another thing to try is to disable call waiting on the
> [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing
> what you've asked it to)...
>
Yep, except on the Polycom, we have found no way to disable
call-waiting.
WW
Willy Wo
oth, but 'outgoing' is confirmed broken.
WW
Willy Wouters
ypOne Publishing
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elect '0'. If this reasoning is
correct, then when would one reasonably use '2' and even '3'
etc.
Sorry to be a pest about this.
WW
Willy Wouters
ypOne Publishing
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that the other end must
> be locking to the zaptel's clock or else clock slips will
> occur.
>
> Feel free to correct me if I'm wrong, but I am pretty sure
> I have this right. :-)
>
> Regards,
> Andrew
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x Brett
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> http://www.loho.co.uk/
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>
use, e.g.
> exten => s,1,DBPut(inuse/chan${ARG1}=1)
> exten => s,2,Dial(SIP/att-${ARG1})
> however, I do not seem to be able to catch the event wich
> releases the channel in order to reset the DB variable.
> exten => h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets
>
n use, e.g.
exten => s,1,DBPut(inuse/chan${ARG1}=1)
exten => s,2,Dial(SIP/att-${ARG1})
however, I do not seem to be able to catch the event wich
releases the channel in order to reset the DB variable.
exten => h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets
executed
Any ideas?
Cheers,
ncel = 64
echocancelwhenbridged = no
echotraining=yes
rxgain => 20%
txgain => -5%
channel => 1
group=2
echocancel = no
signalling=fxo_ks
mailbox = 2100
channel => 2
--
Now it works every time.
Hope this helps
Willy
- Original Message Follows -
&
ts situation.
> > > Especially when there are other issues (zaptel.conf
> > > and zapata.conf) to get a functional system.
> > > Maybe this deserves a wiki input. In any case,
> > > Happy Easter
> > > WW
> > >
> > > - Origina
Easter
> > WW
> >
> > - Original Message Follows -
> > > Dimitri,
> > > I just got off the phone with digium. Here's what I
> > > (from my notes) the event codes mean
> > > Event 4: Alarm detected
> > > Event 5: Alarm cleared
> Event 8: Bad HCS
> > The 6 & 8 which occur sporadically are possibly causing
> > the observed symptoms.
> > Now ... what causes 6 & 8 is the question.
> > Interrupt conflicts was one suggested possibility.
> > Another possibility is 'stuff' fr
co which is not
> understood / mis-understood by the driver.
> I'll keep the list posted.
> Willy
>
> - Original Message Follows -
> > Dear Willy
> > i notice the same problem with my E100P using the
> > latest cvs zaptel driver i have try every t
6 & 8 is the question.
Interrupt conflicts was one suggested possibility. Another
possibility is 'stuff' from the Telco which is not
understood / mis-understood by the driver.
I'll keep the list posted.
Willy
- Original Message Follows -
> Dear Willy
> i not
on
channel 11
Apr 8 17:49:18 WARNING[-1210963024]: PRI: Read on 32
failed: Unknown error 500
Apr 8 17:49:18 NOTICE[-1210963024]: PRI got event: 5 on
span 1
Can someone PLEASE interpret these messages?
BTW: Calls & messages to digium support have gone unanswered
:(
TIA
Willy
- Original M
OOPS ...
> > Just add a ",r" option to the Dial statement, or, do
I get it ;)
Thnx, WW
Willy Wouters
ypOne Publishing
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More Info:
And I went back to CVS-03/26/04 and can hear the 'ringing'
again when I call in to the box ...
BTW: This behavior exists on the production system (T1 PRI
interface to PSTN only) and on the Developent system
(FXO/FXS and IAX2 interfaces)
Cheers,
Willy
- Original Messa
All,
I upgraded to the [*] stable release branch.
When I call into the box (confirmed on 2 installations) the
caller no longer hears the ringing. The CLI confirms that
extensions are being 'rung'.
Whassup?
Willy
Willy Wouters
ypOne
Thanks,
> Brian
> Brian,
>
> At the CLI, type 'show application VoiceMailMain'. You
> can use the CLI 'show applications' command to list all
> available apps. If you hit tab, it acts just like BASH's
> auto complete. Wonderful feature!
>
&
and then CONFIRM that is what I want the mailsystem to use
OR CANCEL out and leave things the way they are?
The same goes, of course, for the unavailable and busy
greetings
TIA
Willy
Willy Wouters
ypOne Publishing
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That's what the 'silence' files were invented for.
See loligo.com (forgot the exact reference, but do a wiki
for J Todd's sound files).
Yes, it's a hack, but it works.
Cheers,
Willy
- Original Message Follows -
> Greetings,
>
>
>
> Below is pa
Works like a charm for me.
I have both VoicePulse and NuPhone registered in IAX.
Depending upon the phone nr dialed, I send a call via NP or
VP.
And yes, my [*] box is behind a NAT.
Include the relevant lines of your iax.conf so we can take a
look.
Cheers, Willy
- Original Message Follows
in the phone config
Cheers, Willy
- Original Message Follows -
> Hi there,
> I am still trying to make the asterisk SIP proxy server
> work with my Grandstream 100 IP phones.
> I tried Stephen advice and it did not work. I stil got the
> 404 error message. So, rigth now,
t the hardware design etc. BTW, I tried
kewlstart, loopstart etc. and it doesn't make any
difference. As I said, it's intermittent on POTS, and it's
constant on my ISDN fxs channels.
Cheers,
Willy
- Original Message Follows -
>
> I am using the wildcard X100P with *. P
k
> Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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; it. Thanks.
>
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> The most personalized portal on the Web!
>
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cret = 2200
host = dynamic
dtmfmode = rfc2833
context=intern
mailbox = 2200
In extensions.conf I have
exten => 2200,1,Dial(SIP/2200,20,tT)
Now, [*] is at 192.168.1.16. Where does the 'header' you
refer to get sent?
I tried adding intercom=true to the sip.conf but that is
feature? How can I turn it on?
>
> Thanks a lot,
> Gelson
>
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l filtering.
> >
> >You can probably blame me for the original switch of the
> Reply-To >header. I believe I am the one who requested it
> soo long ago. >--
> >Steven Critchfield <[EMAIL PROTECTED]>
> >
> >______
; Company
> > directory list exten => 9,1,Goto(npitest|s|1);
> > VoIP Testing Menu
> > Rich
> >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://
].
Anyone who has this successfully working with snom, please
respond .. Using the [*] sound card for a separate PA
system is NOT an option ;)
As I said, I will be 'distilling' the info and turn it into
a wiki entry.
Cheers and TIA,
Willy
Willy Wouters
ypOne Publishing
__
econds the snomm 200 beeps the MWI goes
> on the LCD and the light flashes a call from asterisk "Not
> Found"
>
> Willy if you could let me see you sip and config files, if
> you have yours working? I'm very sure it is not a LAN
> issue, but a config issue
>
> t
dial [EMAIL PROTECTED],
> I have not exten or
> account "asterisk" ???, can't even find where this is set
> ?
>
> Thanks again
> Barry
>
>
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1 (which I
assume you are using) is 'registered'?
Willy
- Original Message Follows -
> Greetings All
>
> I'm busy trying out my new snom 200(s)
> I have it connected and * CLI tells me registered
>
> 1) I pick up the handset and hear the dial tone
> 2)
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Willy Wouters
ypOne Publishing
at the Plantronics design specs, there is no
telling whether it would work. In any case, the real answer
is to fix the problem right at the X100P interface.
Cheers,
Willy
- Original Message Follows -
> Hello all,
>
> I'm thinking about getting the Plantronics DSP-400 headset
Yep,
I think it's possible a card / driver issue.
I tested on POTS (Alltel communications - Texas) and the
behavior did not change. Wonder if digium still monitors
this list.
Cheers,
Willy
- Original Message Follows -
> [EMAIL PROTECTED] wrote:
>
> > It appears that the
Load FXS device (a TDM400P) as Channel #, and use
kewlstart FXO signalling
fxoks=2
# Load FXO device (a T100X) as Channel #, and use kewlstart
FXS signalling
fxsks=1
-
Verry simple setup. Mostly works, but no 'hangup'
signaling.
Cheers,
Willy
Original
Ahaa!
I am using a line coming out of an ISDN breakout box ..
I'll try it with a regular analog line next.
I'll let you all know what happens.
Thanks for the hint,
Willy
- Original Message Follows -
>
> What sort of phone line are you using? Connecting an
> X100P to
x27; tone. The 'hangup' event is detected.
I searched the archives, but could not find a solution.
Any ideas,
TIA
Willy
Willy Wouters
ypOne Publishing
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e(now disconnected) user to
select an option.
What gives?
Cheers,
Willy
- Original Message Follows -
> Hi all,
>
> if i answer a call on my astbox and go into an AGI
> script... then there is somthing happens.(play music or
> something like that)...and the person who called to th
Hi all!
Having stuck my neck out and going for [*] for our new
sales office, (instead of upgrading a
Meridian-Nortel key system), one of the main concerns
remains the support (or lack thereof) of paging /
intercom functionality. Maybe I am just missing somthing
elementary? Here goes:
After p
Hi ..
When the receptionist parks a call for someone who is in the
building
but not at their desk, she does an 'allpage' which
blares-out over
the intercom system: 'Willy you have a call parked at 101'.
Willy can then just grab any phone (kitchen, hall, computer
room)
and pic
Monastery is neat as a monitoring tool. The console's we're
talking
about also let the user pick-up calls etc.
- Original Message Follows -
> See monastery, maybe help you
> (http://pbx.unslept.com/newstatus.php)
>
> Regards,
>
> Gus
>
> - Original Message -
> From: <[EMAIL P
Hi All!
I am thinking about fork-lift-upgrading a Nortel-Meridian
key system with a * PBX driving SIP phones in the office.
The interface to PSTN would be a fractional T1 PRI (11 lines
plus D channel). The GS phones look acceptable for most
users. The forthcoming "Sayson 480i" would work for
manag
Simon,
Do the GS phones support stutter tone as-well-as
the message light?
I am thinking about buying a load of GS-102's
for the office.
Any other comments appreciated.
TIA
Willy
- Original Message Follows -
> Haha
>
> The magic tweak,, I knew there had to be one.
>
= 2, channel = 2
msetup_zap: Unable to register channel '2'
WARNING: mast_load_resource: load_module failed, returning
-1
-
Any ideas?
BTW: trying to make the X100P channel 1 does not help
either. ztcfg shows the channels, but asterisk will fail to
load.
Chee
f
file, but I have no idea what.
Of course, the kit also has a TDM400P card. I have a phone
plugged into it, but (needless to say) there is no dial tone
or anything.
Help Please ...
willy
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