[asterisk-users] how to stop web Click to Call fraud, robots, etc

[Chris Earle]

Hi all,

I'm writing some code to do a web 'click to dial' sort of thing.  Where the
surfer puts in their number and some php/asterisk API code Originates a call
out to them and connects them to an internal extension.


But this raises a number of security/nuissance issues:
I'm well aware that the numbers entered should be validated for local
dialing etc
But...

*What if a robot hits the page, fills out the form with a legit number, and
effectively causes a prank call out to some poor soul?
*invalid area codes? how to deal with?  Check against a list of valid ones?

That's all I can think of right now.  Can all these issues be dealt with by:
1 -- a sort of easy route, add a CAPTCHA to the web form
2 -- compare against lists, or somehow do asterisk dialplan logic to stop
wellhow could you stop legit numbers? :-S

Ideas, suggestions appreciated!!


-- 
--
Chris Earle




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Re: [asterisk-users] Star Wars Echo Sound

[Chris Earle]

I wanna say that's the echotraining taking effect.

What it does is try to cause some echo so it can dynamically reconfigure the
levels on the fly -- right at the start of the call.  I know this happens
with digium cards -- not sure if the Sangoma cards behave the exact same
way.  It's only at the start of the call right? once that occurs, the EC is
kicked in and everything is fine?

--
Chris Earle
System Solutions Specialist,
Network Technologies Division

CBL Data Recovery
w: http://www.cbltech.com



"Rob Schall" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> We have a location that is having a really odd issue. We have a sangoma
> POTs card. We are running software echo cancellation with the card
> (through asterisk) to try to eliminate some major echoing problems. I've
> turned on both EC and echotrain, which seemed to have gotten rid of the
> echo for the most part. However, we are now running into an issue where
> the outside caller hears a star wars type of sound. I expierenced this
> myself when talking to them. By this, I mean you hear a few words from
> them, then a few seconds lagging behind, you'll hear a muffled (darth
> vader) version of the same thing.
>
> Has anyone seen this?
> Thanks,
> Rob
>
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[asterisk-users] Passing variables over IAX2 -- IAXVAR patch?

[Chris Earle]

Hi all,

maybe someone on the dev list can jump in here too

I have run into the wellknown problem with IAX about passing variables
between servers.  In my current 1.0.x network with IAX peers and Queues -- 
passing some extra variable information between Queue member channels is
obviously not going to work.

Googling shows some old discussions of an IAXVAR patch ..but I can't
seem to find it anywhere.  Anyone know about this or if the solution worked?
Discussions seem to indicate that it worked .as a sort of temp fix for
those that needed it :-S

Any help/suggestions appreciated

-- 
--
Chris Earle
System Solutions Specialist,
Network Technologies Division


CBL Data Recovery
Markham Facility

http://www.cbltech.ca


Be committed to the environment. Please think twice before printing this
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Re: [asterisk-users] Queue calls drop to voicemail intermittantly

[Chris Earle]

so I tried this ...and locally on the same server, the channel variable
'VMFLAG' works great -- gets checked for direct calls, and set to 0 when
sending calls through Queues.

But the power of Chan Local/ to  send calls between multiple servers is
ruined because now if you dial direct a Local/ext  ... it won't pass that
variable along so all calls to external channels are considered Queue
calls.

:-\

gotta maintain that 'information' when I dial out over IAX2 to the other
peer


any random ideas appreciated,

--
Chris


"Chris Earle" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> To solve the problem of Local channels answering Queue calls, I thought
> about myabe using a channel variable switch that turns on before the Queue
> is called, and a check in the extension-dial to Local/ext to see if it is
a
> queue call and shouldn't go to voicemail, or if it's just a direct call
that
> should go to voicemail after a certain time. ...I think it'll work,
> though I haven't tried it yet
>
> --
>
> "Tilghman Lesher" <[EMAIL PROTECTED]> wrote in message
> news:[EMAIL PROTECTED]
> > On Monday 17 December 2007 12:35, Gregory Malsack wrote:
> > > Can anyone tell me what might cause callers on hold in a queue to drop
> > > into agents voicemail boxes?
> >
> > Probably you're putting "Local" channels into the queue.  Any answer
event
> at
> > all generated by the Local channel, including one generated by
Voicemail,
> is
> > considered a pickup by the Queue app.  Note that if you use the raw
> channel
> > (SIP/IAX/Zap/whatever), then this will not happen when a queue member
> fails to
> > answer their phone.
> >
> > Or create extensions that do not end in Voicemail for the use of Queue.
> >
> > -- 
> > Tilghman
> >
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Re: [asterisk-users] Queue calls drop to voicemail intermittantly

[Chris Earle]

To solve the problem of Local channels answering Queue calls, I thought
about myabe using a channel variable switch that turns on before the Queue
is called, and a check in the extension-dial to Local/ext to see if it is a
queue call and shouldn't go to voicemail, or if it's just a direct call that
should go to voicemail after a certain time. ...I think it'll work,
though I haven't tried it yet

--

"Tilghman Lesher" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On Monday 17 December 2007 12:35, Gregory Malsack wrote:
> > Can anyone tell me what might cause callers on hold in a queue to drop
> > into agents voicemail boxes?
>
> Probably you're putting "Local" channels into the queue.  Any answer event
at
> all generated by the Local channel, including one generated by Voicemail,
is
> considered a pickup by the Queue app.  Note that if you use the raw
channel
> (SIP/IAX/Zap/whatever), then this will not happen when a queue member
fails to
> answer their phone.
>
> Or create extensions that do not end in Voicemail for the use of Queue.
>
> -- 
> Tilghman
>
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[asterisk-users] How to use AddQueueMember with IAX2 peers?

[Chris Earle]

Hi all,

I've been working on this for days and can't find a solution.  I need to use
AddQueueMember for my agent logins to my Queues -- but a number of my agents
are outside the main server, which is connected to my asterisk network over
IAX2.  I can't just do a AddQueueMember(queuename) because it puts in a
complicated member calleridnum like: IAX2/peername:65723/23
Which won't exist when it comes time to transfer a call to that member.

Help!

I have tried using the chan_local formatted strings instead  like
Local/[EMAIL PROTECTED] -- but you lose all sorts of functionality if
you do it that way



-- 
--
Chris




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Re: [asterisk-users] Queue members, URI.

[Chris Earle]

Hi all,

sorry to rehash this - but I'm having similar issues.  I'm on Asterisk 1.0
and have been using Queues without any problems locally.  I mean, all the
SIP devices on my local server can be added to queues using AddQueueMember.
However, I now need to allow agents from other servers to log in to the
queue and I thought I could do this with IAX2/calleridnum or something
..but it doesn't work.  The only way I was able to get it to work was by
defining them as Local/@context
But this has major drawbacks.  They are in the queue and can receive
calls -- but when the queue directs a call to them, it loses control over it
and calls are just transfered to the one agent and don't timeout the
caller in the queue isn't really in the queue anymore...

The reason it didn't work with IAX2 was that every time an agent logged in
... Add QueueMember would put them in as IAX2/iaxpeer/ ...
because that's where they were connecting over at that very moment.  But the
queue is unable to locate them at that same port when an actual call comes
into the queue!  Since they are always moving around ports under the IAX2
protocol.

So using Local works cause it uses the dialplan's intelligence in locating
an extension on an iaxpeer -- but it's not really a channel like Zap or Sip
... so queue functionality is lost

So I'm revisiting this now --- is there any way to use IAX2 peers as queue
members?  Maybe I'm writing the URI's wrong
Or is this something that has been fixed drasically in asterisk 1.2/1.4
anyone know?

Ideas/suggestions appreciated ...

--
Chris Earle


"Thomas Kenyon" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Is there an advantage to having a Queue members URI in the form:
>
> SIP/User  (or indeed IAX2/User)
> Over
> Local/@context
>
> ?
>
> I know that the latter will allow you to do things like set counting
> logic etc. through dialplan operations, but the former appears to be a
> more direct route to calling the party. (and if need be, there is the
> ability in queues to run a script on connection iirc).
>
> TIA for any clarification.
>
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Re: [asterisk-users] Pickup command

[Chris Earle]

Not really sure about SIP exactly, but for Asterisk 1.0 versions, I know
that the Pickup only works with Zaptel channels -- so to use it for any sort
of IP channel, IAX for example, you have to use an addon/patch  google
it, 'pickup2' I think it's called works well, allows the Pickup
command to grab any ZAP or IAX channel

--
Chris Earle


"Carlos Chavez" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
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Re: [asterisk-users] Converting an audio file to a ".gsm" format

[Chris Earle]

Digium has a handy tool online!
http://www.digium.com/en/products/voice/audioconverter.php

:-)

--
Chris

"Alex Balashov" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On Sun, 12 Aug 2007, MOSBAH ABDELKADER wrote:
>
> > Hello all,
> >
> > have anyone an idea about converting an audio file (.wav, .mp3, .au,...)
to
> > a ".gsm" audio file to use it as a voicemail file with Asterisk.
>
>'sox' should be able to do this, AFAIK.
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: +1-678-954-0670
> Direct : +1-678-954-0671
>
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Re: [asterisk-users] Parsing IAXPeers from Asterisk Manager(PHP API)

[Chris Earle]

Great idea/strategy, thank you!

Going to be c00l if I can get this working

regards,

--
Chris


"Jared Smith" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On Thu, 2007-07-19 at 09:16 -0400, Chris Earle wrote:
> > Trying to do what should be a basic info retrieval from my asterisk
> > network -- get the list of IAXPeers and break down by IP, name, and
Status
> > etc.
>
> Unfortunately, this is one of those cases where the AMI output is the
> same as the CLI output, which makes it difficult to parse.  (Regular
> expressions can help, but you obviously have an aversion to them, so
> we'll use another method.) As long as your IAX peer names are shorter
> than 16 characters, you can use a trick I've used for a while... first
> get the list of iax2 peer names:
>
> Action: IAXPeers
> ActionID: 987325918659161
>
> ActionID: 987325918659161
> Name/UsernameHost Mask Port
> Status
> flibbertygibbit  (Unspecified)   (D)  255.255.255.255  0
> Unmonitored
> demo/asterisk216.207.245.47  (S)  255.255.255.255  4569
> Unmonitored
> 2 iax2 peers [0 online, 0 offline, 2 unmonitored]
>
> and break on the first space or forward slash.  Once I have an array of
> IAX peer names, I use the Command action to run "iax2 show peer
> " on each of the IAX peers, like this:
>
> Action: Command
> Command: iax2 show peer flibbertygibbit
> ActionID: 762315185916555
>
> Response: Follows
> Privilege: Command
> ActionID: 762315185916555
>
>
>   * Name   : flibbertygibbit
>   Secret   : 
>   Context  : blah
>   Mailbox  :
>   Dynamic  : Yes
>   Callerid : "" <>
>   Expire   : -1
>   ACL  : No
>   Addr->IP : (Unspecified) Port 0
>   Defaddr->IP  : 0.0.0.0 Port 4569
>   Username : hullabaloo
>   Codecs   : 0xe703 (g723|gsm|g729|speex|ilbc)
>   Codec Order  : (none)
>   Status   : Unmonitored
>   Qualify  : every 6ms when OK, every 1ms when UNREACHABLE
> (sample smoothing Off)
>
> --END COMMAND--
>
> That output is obviously somewhat easier to parse, although it's not
> perfect either.  Hopefully the Asterisk development community will
> eventually get around to rewriting much of the AMI actions to make their
> output easier for programs to parse.
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
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[asterisk-users] Parsing IAXPeers from Asterisk Manager (PHP API)

[Chris Earle]

Hi all,

Trying to do what should be a basic info retrieval from my asterisk
network -- get the list of IAXPeers and break down by IP, name, and Status
etc.

Asterisk 1.0 Manager unfortunately.  Using PHP.
Easy to get the response -- but parsing it is impossible! That table format
throws me for a loop ...
Any idea of an easy way to seperate out the columns of that horrid 'table'
reponse layout?
I really want to get the value of the Status ping time ... hope to
produce a neat little google maps mashup out of it.

Parsing ideas appreciated ... go lightly on the regex if possible! ;-)



--
Chris




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[asterisk-users] Re: Multiple TDM400p cards in one machine -- nolonger an issue?

[Chris Earle]

Well, yeah, I know it's do-able with either the Sangoma card or Digium's own
TDM2400  but I don't want to replace the TDM400p I've already got in
there

Anyone think two TDM400p's won't cause me any trouble?

--
Chris



"Lee Jenkins" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Chris Earle wrote:
> > Hi all,
> >
> > Years ago, I was pretty sure attempting to use two TDM400p cards in one
> > machine was recommended against by Digium ... probably because the cards
> > couldn't hack it, and/or interrupt problems etc
> >
> > I have seen some posts recently that seem to indicate it is in fact
possible
> > these days thanks to some updated firmware perhaps? . I just need to
> > have two in the server because the 4 ports aren't enough ...
> >
> > I'd rather just expand by one card rather than get a TDM2400 (or
TDM800??)
> >
> > Anyone had recent success/failure with this sort of thing?
> >
> >
>
> Sangoma Remora Card may be an option?
>
> http://www.sangoma.com/datasheets/p_a200-specs
>
> -- 
>
> Warm Regards,
>
> Lee
>
>
>
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[asterisk-users] Multiple TDM400p cards in one machine -- no longer an issue?

[Chris Earle]

Hi all,

Years ago, I was pretty sure attempting to use two TDM400p cards in one
machine was recommended against by Digium ... probably because the cards
couldn't hack it, and/or interrupt problems etc

I have seen some posts recently that seem to indicate it is in fact possible
these days thanks to some updated firmware perhaps? . I just need to
have two in the server because the 4 ports aren't enough ...

I'd rather just expand by one card rather than get a TDM2400 (or TDM800??)

Anyone had recent success/failure with this sort of thing?


--
Chris Earle



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[asterisk-users] Re: UK zaptel and zapata.conf for TDM400P

[Chris Earle]

2 TDM400P's in one machine can co-exist?  I thought this was near impossible
 anyone confirm?


--
Chris



"Gordon Henderson" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On Sat, 21 Apr 2007, Steve Kennedy wrote:
>
> > Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's
> > working with UK set-up.
> >
> > They're set-up with 7 analogue phones and 1 PSTN port.
> >
> > Currently zaptel.conf has
> > fxoks=1-7
> > fxsks=8
> > loadzone=uk
> > defaultzone=uk
> >
> > It's really zapata.conf that would be useful.
> >
> > Currently using the zaptel/asterisk that comes with Ubuntu (latest)
> > which needed a bit of tweaking (1.2.16), but could compile latest 1.4
> > release.
>
> See below for a system I'm using right now. 2 PSTN ports and 1 analogue
> port. This is a compile & build from scratch system, not supplied in a
> package or using trixbox, etc.
>
> Curious about your 2 x TDM400 cards though - I'm presuming you've got no
> interrupt issues, etc. ?
>
> ...
>
> Based on this zaptel.conf:
>
>fxoks=1
>fxsks=3
>fxsks=4
>loadzone=uk
>defaultzone=uk
>
> I have:
>
> [trunkgroups]
>
> [channels]
>
> ; Default settings applicable to all channels
>
> usecallerid=yes
> cidsignalling=v23
> cidstart=polarity
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echotraining=yes
> echocancelwhenbridged=yes
> immediate=no
> faxdetect=no
>
> ; Channel 1: Local analogue line
> context=internal
> group=0
> signalling=fxo_ks
> sendcalleridafter=2
> rxgain=0
> txgain=0
> mailbox=103
> callerid=Analogue Port <103>
> channel => 1
>
> ; Channel 3: PSTN line
> context=incoming
> group=1
> usecallerid=yes
> faxdetect=none
> signalling=fxs_ks
> rxgain=7
> txgain=7
> callerid=asreceived
> channel => 3
>
> ; Channel 4: PSTN line
> context=incoming
> group=1
> usecallerid=yes
> faxdetect=none
> signalling=fxs_ks
> rxgain=8
> txgain=8
> callerid=asreceived
> channel => 4
>
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[asterisk-users] Re: qozap: t3 timer expired for span ...

[Chris Earle]

This problem / messages has not gone away ...

No one's got any ideas or explanation about what the card is trying to tell
me?

--
Chris


"Chris Earle" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> bristuff-0.2.0-RC8s
>
> two isdn lines plugged into first two ports
>
> and like I said, also a digium tdm400 card in there for analog phones
>
> this 'timer' error message  it is something to do with the qozap
driver
> isn't it?  not sure
>
> Thanks for any ideas!
>
> --
> Chris
>
>
> "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote in message
> news:[EMAIL PROTECTED]
> > On Thu, Mar 15, 2007 at 10:30:23AM -0500, Chris Earle (CBL) wrote:
> > > Hi all
> > >
> > > message:
> > > qozap: t4 timer expired for span 2
> > > qozap: t4 timer expired for span 3
> > > qozap: t3 timer expired for span 2
> > > qozap t3 timer expired for span 3
> >
> > Which version is it of bristuff?
> >
> > >
> > >
> > > wow -- what does this mean!?  all of a sudden showing up on my server
> ... no
> > > change after reboot ..  Junghanns QuadBRI card in place
> >
> > Anything connected to it? Where exactly?
> >
> > >
> > > affecting outgoing faxing?! (between bridged TDM400 analog card and
> QuadBRI)
> > >
> > > Not a clue why this is .. incoming/outgoing voice calls work,
> incoming
> > > faxes even work but when outgoing fax is dialed, says no one is
> availale
> > > to answer at this time 
> > >
> > > The error has not ever been there before and as far as I know, no isdn
> > > wiring has been changed or anything
> > >
> > > ideas, appreciated!
> >
> > -- 
> >Tzafrir Cohen
> > icq#16849755jabber:[EMAIL PROTECTED]
> > +972-50-7952406   mailto:[EMAIL PROTECTED]
> > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Re: qozap: t3 timer expired for span ...

[Chris Earle]

bristuff-0.2.0-RC8s

two isdn lines plugged into first two ports

and like I said, also a digium tdm400 card in there for analog phones

this 'timer' error message  it is something to do with the qozap driver
isn't it?  not sure

Thanks for any ideas!

--
Chris


"Tzafrir Cohen" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On Thu, Mar 15, 2007 at 10:30:23AM -0500, Chris Earle (CBL) wrote:
> > Hi all
> >
> > message:
> > qozap: t4 timer expired for span 2
> > qozap: t4 timer expired for span 3
> > qozap: t3 timer expired for span 2
> > qozap t3 timer expired for span 3
>
> Which version is it of bristuff?
>
> >
> >
> > wow -- what does this mean!?  all of a sudden showing up on my server
... no
> > change after reboot ..  Junghanns QuadBRI card in place
>
> Anything connected to it? Where exactly?
>
> >
> > affecting outgoing faxing?! (between bridged TDM400 analog card and
QuadBRI)
> >
> > Not a clue why this is .. incoming/outgoing voice calls work,
incoming
> > faxes even work but when outgoing fax is dialed, says no one is
availale
> > to answer at this time 
> >
> > The error has not ever been there before and as far as I know, no isdn
> > wiring has been changed or anything
> >
> > ideas, appreciated!
>
> -- 
>Tzafrir Cohen
> icq#16849755jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] qozap: t3 timer expired for span ...

[Chris Earle \(CBL\)]

Hi all

message:
qozap: t4 timer expired for span 2
qozap: t4 timer expired for span 3
qozap: t3 timer expired for span 2
qozap t3 timer expired for span 3


wow -- what does this mean!?  all of a sudden showing up on my server ... no
change after reboot ..  Junghanns QuadBRI card in place

affecting outgoing faxing?! (between bridged TDM400 analog card and QuadBRI)

Not a clue why this is .. incoming/outgoing voice calls work, incoming
faxes even work but when outgoing fax is dialed, says no one is availale
to answer at this time 

The error has not ever been there before and as far as I know, no isdn
wiring has been changed or anything

ideas, appreciated!


--
Chris Earle
System Solutions Specialist


-- 
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[asterisk-users] Re: FXS - Init Indirect Registers UNSUCCESSFULLY.

[Chris Earle]

ahh! I am having this same problem all of a sudden

I've installed many TDM cards before ..never had this problem

what gives?

Trying to load zaptel 1.0.10  ...

Rev. G card ... tried uncommenting the revH fix in zconfig.h ...but no go

ideas?!

--
Chris



"Michael C. Cambria" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
>
> I'm having problems with a TDM22B.  The FXO modules work fine.  Both FXS
> modules fail to initialized.
>
> The error messages I seeR when the module loads:
>
> Init Indirect Registers UNSUCCESSFULLY.
> Indirect Registers failed verification.
>
> I already RMA'ed a TDM22B because of this problem.  Now that the
> replacement shows a similar issue, I want to see if anyone else can
> think of something to try; at least until Monday when I can get an RMA
> number for this card.
>
> If it helps, I have a TDM20B (just FXS modules) that does not see this
> problem when I place it in the same slot.
>
>
> Here is what dmesg shows for the TDM22B when the system boots, or when I
> issues modprobe wctdm:
>
> Freshmaker version: 73
> Freshmaker passed register test
> !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000
> !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600
> !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000
> !!! COMMON_MAX_TRES iREG 1F = 0 should be 200
> !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0
> !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600
> !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80
> !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000
> !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320
> !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C
> !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100
> !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10
> !!! CM_BIAS_RINGING iREG 28 =  should be C00
> !!! DCDC_MIN_V iREG 29 =  should be C00
> !!! DCDC_XTRA iREG 2A =  should be 1000
> !!! LOOP_CLOSE_TRES_LOW iREG 2B =  should be 1000
> ! Init Indirect Registers UNSUCCESSFULLY.
> Indirect Registers failed verification.
> !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000
> !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600
> !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000
> !!! COMMON_MAX_TRES iREG 1F = 0 should be 200
> !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0
> !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600
> !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80
> !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000
> !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320
> !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C
> !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100
> !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10
> !!! CM_BIAS_RINGING iREG 28 =  should be C00
> !!! DCDC_MIN_V iREG 29 =  should be C00
> !!! DCDC_XTRA iREG 2A =  should be 1000
> !!! LOOP_CLOSE_TRES_LOW iREG 2B =  should be 1000
> ! Init Indirect Registers UNSUCCESSFULLY.
> Indirect Registers failed verification.
> Module 0: FAILED FXS (FCC)
> !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000
> !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600
> !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000
> !!! COMMON_MAX_TRES iREG 1F = 0 should be 200
> !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0
> !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600
> !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80
> !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000
> !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320
> !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C
> !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100
> !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10
> !!! CM_BIAS_RINGING iREG 28 =  should be C00
> !!! DCDC_MIN_V iREG 29 =  should be C00
> !!! DCDC_XTRA iREG 2A =  should be 1000
> !!! LOOP_CLOSE_TRES_LOW iREG 2B =  should be 1000
> ! Init Indirect Registers UNSUCCESSFULLY.
> Indirect Registers failed verification.
> !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000
> !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600
> !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000
> !!! COMMON_MAX_TRES iREG 1F = 0 should be 200
> !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0
> !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600
> !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80
> !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000
> !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320
> !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C
> !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100
> !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10
> !!! CM_BIAS_RINGING iREG 28 =  should be C00
> !!! DCDC_MIN_V iREG 29 =  should be C00
> !!! DCDC_XTRA iREG 2A =  should be 1000
> !!! LOOP_CLOSE_TRES_LOW iREG 2B =  should be 1000
> ! Init Indirect Registers UNSUCCESSFULLY.
> Indirect Registers failed verificat

[asterisk-users] Re: Jabber/Asterisk Integration

[Chris Earle]

"agent monitoring screen"?

curious,
which app are you using for that?

--
Chris Earle


"Julian Lyndon-Smith" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
>
> Kyle Sexton wrote:
> > Started playing with 1.4 and I'm curious what uses people have come up
> > with for the Jabber integration?  So far I can think of presence based
> > call routing, but I'm sure there are other ideas.  How are YOU using
> > the new Jabber features in 1.4? :)
> >
>
> We've been using it since July last year (brave / stupid - make your
> choice) for integrating our custom application with the asterisk system.
> The phone system sends all sorts of call information to the agent about
> to receive the call, whilst the agent monitoring screen is used to
> monitor the presence of the agents and their dialplan status (dialling /
> calling / etc etc)
>
> Julian.
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[asterisk-users] receiving fax with junghanns quadbri bristuff

[Chris Earle]

Hey,

anyone know if it's possible to receive faxes through a Junghanns bristuff
quadbri card?

In germany, currently I have faxes coming in on DID line into QuadBRI and
then passing to Digium TDM400 (analog) and into faxmachine.  But the
reliability of TDM card is spotty, so I want to maybe just accept faxes in
on ISDN card and save on asterisk system ...?  keeping digital signal strong
...


ideas appreciated!!

-- 
--
Chris Earle
System Solutions Specialist,



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[asterisk-users] Re: Disconnect Supervision UK / BT solution?

[Chris Earle]

Wondering if you ever got this change made and if it did anything?

Update us if you please :-)

--
Chris



"Chris Earle (CBL)" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> thanks for your helpful investigation!  I await news :-)
>
> --
> Chris
>
>
> - Original Message - 
> From: "Matt Brown" <[EMAIL PROTECTED]>
> To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing
List -
> Non-Commercial Discussion" 
> Sent: Saturday, January 20, 2007 7:55 AM
> Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution?
>
>
> > Well,
> >
> > I have just phoned BT today who said they can increase the CPC value
> > on the line - however it needs to be done at the exchange - and has
> > been booked for Tues.
> >
> > I suppose I will know wether this worked on Tues :-) - I shall post
> > my findings.
> >
> > Regards
> >
> > --
> > Matt Brown
> >
> >
> >
> > On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote:
> >
> > > Hi all
> > >
> > > I'm using sangoma a200 cards in the UK and have the ongoing, often
> > > noted
> > > problem of disconnect supervision with BT POTS lines.
> > >
> > > Just noticed this post on
> > > http://www.voip-info.org/wiki/view/UK+Asterisk+Details
> > > stating that potentially someone's got a solution :
> > >
> > > "TDM400P & Not Detecting Hangups:
> > >
> > >  Got a TDM400P installed and having problems with Asterisk not
> > > detecting
> > > hangups? Using BT? If so, contact BT and ask what the "Disconnect
> > > Clear
> > > Time" setting is for your phone line. Odds are it's probably 100.
> > > Increasing
> > > it to 800 fixed the issue for me.
> > >
> > > "Disconnect Clear Time" is BT's name for CPC. "
> > >
> > >
> > > Does anyone have any thoughts/confirmation about this finally being
> > > a viable
> > > solution?  This disconnect supervision problem has plagued TDM and
> > > Sangoma
> > > cards for a long time!
> > >
> > > Comments appreciated before I get on the phone with BT
> > >
> > >
> > > --
> > > Chris Earle
> > > System Solutions Specialist
> > >
> > >
> > > -- 
> > > This message has been scanned for viruses and
> > > dangerous content and is believed to be clean.
> > >
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> > > asterisk-users mailing list
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> >
> >
> > -- 
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> > dangerous content and is believed to be clean.
>
>
> -- 
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[asterisk-users] Re: On what distribution is www.asterisknow.com basedon ?

[Chris Earle]

I'm tempted to rebuild my asterisk network with AsteriskNow - my
question is, can you ADD anything to it?  i.e. cdr_mysql logging?  I thought
I saw it didn't have that

And how does it handle the hardware?  I don't use digium cards in all of my
servers because of country issues (Junghanns in Germany, Sangoma in UK, etc)

If it's expandable through usual package addons etc, then it would seem
there is alot of added value because of the increased EASE of administration
over your well-maintained debian box 

Thoughts?

--
Chris


"Maxim Veksler" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Hello Asteriskies,
>
> Has someone tried www.asterisknow.com ?
>
> What is the package manager used? And what is the added value compared
> to the well maintained debian based asterisk ?
>
> Thanks,
>
>
> -- 
> Cheers,
> Maxim Veksler
>
> "Free as in Freedom" - Do u GNU ?
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[asterisk-users] Re: kewlstart disconnect threshold

[Chris Earle]

Can anyone confirm that it actually works in Singapore with Busy Detect?

I have a system with loopstart and BusyDetect and have recently attempted to
improve disconnect detection results with the addition of
hanguponpolarityswitch ... results are mixed


--
Chris Earle
System Solutions Specialist



"Leo Ann Boon" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Stephen Bosch wrote:
> > The reason we have these complaints is not because Asterisk doesn't
> > detect the drop -- it's because a great many telephone companies don't
> > do remote party disconnect signalling, or they don't do it properly.
> > When people call for technical assistance they usually end up talking
> > with someone who has no idea what Calling Party Control or remote party
> > disconnect actually is.
> >
> > Case in point:
> >
> > I am with Telus in Alberta, Canada. By default, the "loop drop" (it's
> > actually a battery drop, as near as I can tell, but kewlstart will
> > detect both) occurs after more than a minute. On some lines it doesn't
> >
> A minute is rather long. CPC when working should be almost immediate
> (see Mark's kewlstart test). What happens if you change it to loopstart?
> Does asterisk detect the drop?
>
> My analog line with Singapore Starhub does a battery drop after about
> 90s (i can hear a short crackling sound), but it's not detectable (no
> console message with verbose >6)  unless I set to kewlstart. Not even
> with hanguponpolarityswitch=yes. So, I guess zaptel doesn't do anything
> if you set signaling to loopstart.
>
> The long and short of it, busydetect maybe the best solution to force
> asterisk to hangup after the remote party hung up.
>
> Leo
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[asterisk-users] Re: Re: SIP "Lines" Example Citel

[Chris Earle]

I'm in the middle of trying to solve that very problem -- integrate a legacy
pbx into my asterisk/IP network.  (Norstar MICS for the record)

Been mulling over how to do it with Quintum boxes the last few days --- very
complicated.  The alternative solution is to use a Citel gateway with the
30+ handsets we have and scrap the MICS pbx and build an asterisk box with
our incoming POTS lines into a digium/sangoma card.

Any testimonials about these approaches? (Quintum boxes or Citel handset
gateways)  Please share!

--
Chris Earle
System Solutions Specialist



"Steve Totaro" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Steven wrote:
> > Setting Realm to asterisk worked for me.
> >
> > ref. from sip.conf:
> > ;realm=mydomain.tld  ; Realm for digest authentication
> > ; defaults to "asterisk". If you set a system name in
> > ; asterisk.conf, it defaults to that system name
> > ; Realms MUST be globally unique according to RFC 3261
> > ; Set this to your host name or domain name
> >
> >
> >
> >
> Yes it worked for me too.  Awesome box if you have to integrate/migrate
> from a legacy system to Asterisk but keep your old handsets.  I am very
> impressed.  The two devices I really like now are the Citel Gatways and
> the Quintum boxes.
>
> Thanks,
> Steve Totaro
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[asterisk-users] Re: Disconnect Supervision UK / BT solution?

[Chris Earle]

Sure -- this is for my Sangoma a200

-=-=-=-
Zapata.conf
(currently supporting CallerID)

[channels]
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0

musiconhold=default

callwaiting=no
transfer=yes
usecallingpres=yes
cadence=200,200,200,4000

usecallerid=yes
callerid=asreceived
cidsignalling=v23
cidstart=polarity
sendcalleridafter=0

;configure incoming lines
group=1
context=UkExt
rxgain=6
txgain=1
signalling=fxs_ks
channel => 1-2,5-6

;configure outgoing lines
group=2
callgroup=2
pickupgroup=2
context=UkInt
rxgain=-1
txgain=2
signalling=fxo_ks
channel => 3-4,7-8
-=-=-=-=-=-=-

-=-=-= Zaptel.conf =-=-=-
fxsks=1-2,5-6
fxoks=3-4,7-8
loadzone = uk
defaultzone=uk





Ideas?

--
Chris



"Carlos Rojas" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
Hello,


What's your zapata.conf and zaptel.conf?




On 1/20/07, Matt Brown <[EMAIL PROTECTED] > wrote:
Well,

I have just phoned BT today who said they can increase the CPC value
on the line - however it needs to be done at the exchange - and has
been booked for Tues.

I suppose I will know wether this worked on Tues :-) - I shall post
my findings.

Regards

--
Matt Brown



On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote:

> Hi all
>
> I'm using sangoma a200 cards in the UK and have the ongoing, often
> noted
> problem of disconnect supervision with BT POTS lines.
>
> Just noticed this post on
> http://www.voip-info.org/wiki/view/UK+Asterisk+Details
> stating that potentially someone's got a solution :
>
> "TDM400P & Not Detecting Hangups:
>
>  Got a TDM400P installed and having problems with Asterisk not
> detecting
> hangups? Using BT? If so, contact BT and ask what the "Disconnect
> Clear
> Time" setting is for your phone line. Odds are it's probably 100.
> Increasing
> it to 800 fixed the issue for me.
>
> "Disconnect Clear Time" is BT's name for CPC. "
>
>
> Does anyone have any thoughts/confirmation about this finally being
> a viable
> solution?  This disconnect supervision problem has plagued TDM and
> Sangoma
> cards for a long time!
>
> Comments appreciated before I get on the phone with BT
>
>
> --
> Chris Earle
> System Solutions Specialist
>
>
> --
> This message has been scanned for viruses and
> dangerous content and is believed to be clean.
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
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[asterisk-users] Re: Disconnect Supervision UK / BT solution?

[Chris Earle]

Yes, IAX <--> IAX works fine

it's when it's Zap(in) ->Asterisk -> Zap(out)


Every call that I pass through to that outgoing Zap channel (a dial out to a
mobile phone) fails to hang up.

CallerID is working -- with this sangoma card, which seems to need...
cidsignalling=v23
cidstart=polarity
...to work in the UK

If removing helps the disconnect detection, I don't mind losing my callerID
support.

--
Chris



"Ed W" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Chris Earle (CBL) wrote:
> > Sorry -- you're right, I didn't express the scenario properly ..
> >
> > The disconnect supervision problem is when I 'forward'/divert an
incoming
> > POTS call out another FXO channel to a mobile phone or POTS line.
> > (POTS -> Sangoma|Asterisk -> POTS/mobile)
> >
> > When the incoming POTS hangs up and/or the mobile the person was
connected
> > to .. Asterisk/Sangoma doesn't hang the Zap channels up.
> >
>
> Just to clarifydoes it all work ok if you are using SIP or IAX for the
> forwarded channels?  Eg local SIP phones?
>
> I only have incoming zap lines in my config and with the exception of
> hangup on ringing I have found hangup detection to work fine.  I have a
> fax machine forwarding in my config as well and again no problems yet
> with hangup on that.
>
> Does it fail to work *every* time, or just intermittently?  Does
> CallerId work ok in your setup?  (can be a clue to help diagnose your
setup)
>
> Ed W
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Re: [asterisk-users] Disconnect Supervision UK / BT solution?

[Chris Earle \(CBL\)]

Sorry -- you're right, I didn't express the scenario properly ..

The disconnect supervision problem is when I 'forward'/divert an incoming
POTS call out another FXO channel to a mobile phone or POTS line.
(POTS -> Sangoma|Asterisk -> POTS/mobile)

When the incoming POTS hangs up and/or the mobile the person was connected
to .. Asterisk/Sangoma doesn't hang the Zap channels up.

I have tried busydetect and busycounts and a number of settings are enabled
for UK CallerID support (polarity switch stuff) ... but I had some sketchy
side effects with busydetect etc and am wary of premature hangups


Thanks for your query

--
Chris



- Original Message - 
From: "Ed W" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Friday, January 19, 2007 1:26 PM
Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution?


>
> > Does anyone have any thoughts/confirmation about this finally being a
viable
> > solution?  This disconnect supervision problem has plagued TDM and
Sangoma
> > cards for a long time!
> >
>
> Just to be clear, what is the exact "disconnect problem" that you see?
>
> I have three TDM cards in two different systems, one using PBX lines and
> one on a private BT line.  Both of them have trouble detecting a caller
> who is ringing, but then hangs up before being answered by the asterisk
> system
>
> However, *all* of them are absolutely fine at spotting a normal hangup
> once the call is connected and I see no random disconnects during calls
> either.
>
> Can you confirm that this is what you mean, or whether it's something
else?
>
> Ed W
>
>
> -- 
> This message has been scanned for viruses and
> dangerous content and is believed to be clean.


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Re: [asterisk-users] Disconnect Supervision UK / BT solution?

[Chris Earle \(CBL\)]

thanks for your helpful investigation!  I await news :-)

--
Chris


- Original Message - 
From: "Matt Brown" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Saturday, January 20, 2007 7:55 AM
Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution?


> Well,
>
> I have just phoned BT today who said they can increase the CPC value
> on the line - however it needs to be done at the exchange - and has
> been booked for Tues.
>
> I suppose I will know wether this worked on Tues :-) - I shall post
> my findings.
>
> Regards
>
> --
> Matt Brown
>
>
>
> On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote:
>
> > Hi all
> >
> > I'm using sangoma a200 cards in the UK and have the ongoing, often
> > noted
> > problem of disconnect supervision with BT POTS lines.
> >
> > Just noticed this post on
> > http://www.voip-info.org/wiki/view/UK+Asterisk+Details
> > stating that potentially someone's got a solution :
> >
> > "TDM400P & Not Detecting Hangups:
> >
> >  Got a TDM400P installed and having problems with Asterisk not
> > detecting
> > hangups? Using BT? If so, contact BT and ask what the "Disconnect
> > Clear
> > Time" setting is for your phone line. Odds are it's probably 100.
> > Increasing
> > it to 800 fixed the issue for me.
> >
> > "Disconnect Clear Time" is BT's name for CPC. "
> >
> >
> > Does anyone have any thoughts/confirmation about this finally being
> > a viable
> > solution?  This disconnect supervision problem has plagued TDM and
> > Sangoma
> > cards for a long time!
> >
> > Comments appreciated before I get on the phone with BT
> >
> >
> > --
> > Chris Earle
> > System Solutions Specialist
> >
> >
> > -- 
> > This message has been scanned for viruses and
> > dangerous content and is believed to be clean.
> >
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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[asterisk-users] Re: Background + Dial

[Chris Earle]

Hmm...I have just noticed this issue as well

I want the background command to play soundfiles while the dialplan moves on
and is dialing a number of zap channels etc...

It plays, but essentially ends up being no different than Playback() 

I note now before posting this, that the Background is intended to ALLOW
caller input (as in an IVR) while sound is playing  I guess I am
confused as to the Background command's purpose ...

Any ideas for what I want to do ?

--
Chris


<[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
Thanks for your reply,

But I want to have an interactive menu, not just a music.
So, the customer can have information menu while he's
waiting the call is answer.
I'dont now if it's possible with MoH.

Thanks a lot

-- Initial Header ---

>From  : [EMAIL PROTECTED]
To  : "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
Cc  :
Date  : Tue, 27 Jun 2006 18:58:26 +0700
Subject : RE: [Asterisk-Users] Background + Dial

Hi GL

Pls. config MOH and use Dial command with "m" option.
This will allow you execute Dial command while providing
Music in the
background.

Hope it help

Hoa Thai Duy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, June 27, 2006 5:01 PM
To: asterisk-users
Subject: [Asterisk-Users] Background + Dial

Hi everybody,

I try this :

[incoming_from_fxo_card]
exten => s,1,Answer()
exten => s,2,Background(filename)
exten => s,3,Dial($(INTERNAL_SIP_TEL))

But * wait the file is finish before make Dial to SIP channel.


Background(filename)  (from voip-info.org) => Starts playing
a given sound
file, but immediately returns, permitting the sound file to
play in the
background while the next commands (if any) execute.

I want to Dial a SIP channel while playing sound and waiting
for a digit
from a ZAP channel. In other words, i want to make a
interactive MoH while
waiting for the SIP channel answer.
Is it possible?

Thanks a lot and excuse me for my poor english (I'll fix
this in few
months).

GL

- ALICE SECURITE ENFANTS
- Protégez
vos enfants des dangers d'Internet en installant Sécurité
Enfants, le
contrôle parental d'Alice.
http://www.aliceadsl.fr/securitepc/default_copa.asp


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- ALICE SECURITE ENFANTS -
Protégez vos enfants des dangers d'Internet en installant Sécurité Enfants,
le contrôle parental d'Alice.
http://www.aliceadsl.fr/securitepc/default_copa.asp


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[asterisk-users] Disconnect Supervision UK / BT solution?

[Chris Earle \(CBL\)]

Hi all

I'm using sangoma a200 cards in the UK and have the ongoing, often noted
problem of disconnect supervision with BT POTS lines.

Just noticed this post on
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
stating that potentially someone's got a solution :

"TDM400P & Not Detecting Hangups:

 Got a TDM400P installed and having problems with Asterisk not detecting
hangups? Using BT? If so, contact BT and ask what the "Disconnect Clear
Time" setting is for your phone line. Odds are it's probably 100. Increasing
it to 800 fixed the issue for me.

"Disconnect Clear Time" is BT's name for CPC. "


Does anyone have any thoughts/confirmation about this finally being a viable
solution?  This disconnect supervision problem has plagued TDM and Sangoma
cards for a long time!

Comments appreciated before I get on the phone with BT


--
Chris Earle
System Solutions Specialist


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[asterisk-users] Re: Disconnect supervision in India?

[Chris Earle]

Thanks for the input

not seeing any evidence of disconnect supervision or callerid  but
you're right, depends on provider ---
Anyone have any luck with provider MTNL?


--
Chris



"Rajkumar S" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> On 1/1/07, ram <[EMAIL PROTECTED]> wrote:
> > On 12/30/06, Rajkumar S <[EMAIL PROTECTED]> wrote:
> > > On 12/29/06, Chris Earle <[EMAIL PROTECTED]> wrote:
> > > > anyone know the status of disconnect supervision on POTS lines in
India?
> > > > Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to
have
> > > > disconnect supervision..
> > >
> > > It does not work afaik, you may not get caller id also. I tested upto
> > > 1.4b3 and no luck.
>
> > its all depends on the provider where you take from.
>
> Does any provider's land line works well with TDM Cards?
>
> raj
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[asterisk-users] Disconnect supervision in India?

[Chris Earle]

Hey all,

anyone know the status of disconnect supervision on POTS lines in India?
Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have
disconnect supervision..

Thanks


--
Chris Earle
System Solutions Specialist



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[asterisk-users] CDR stats to one mysql database, multiple webstats packages

[Chris Earle]

Have a number of asterisk servers and want to get some good stats tracking
going (with Asterisk-Stat) -- but this requires cdr logging to mysql, apache
and the stats software running on each server.

Or does it?  Of course, I can either run the stats package on the webserver
and direct it to each individual server's local mysql db --- or have each
asterisk server logging to an external mysql db somewhere.(on the
webserver I suppose)

Thoughts on this?  Good idea/Bad idea to log to an external source?  One
thing that might be an issue is if for some reason the external source
becomes unreachable or goes offline ...then what happens to the CDR data for
that time period?

Suggestions appreciated

--
Chris



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[asterisk-users] IAX call drops, recent instability

[Chris Earle]

Hi all

I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1,
with TDM cards for analog lines.  They have been in production use for many
months, handling incoming calls, and also allowing daily inter-server calls
over IAX (transfers, extension calls etc)

All of a sudden, in the last 3-4 weeks, with little to no changes to any
config or setup on the servers -- a large number of IAX-IAX calls are
dropping.  It is driving me nuts because I can't pinpoint any change in the
system that might be a catalyst ... nor rectify with any modifications to
iax.conf, zapata.conf etc

All servers are iax.conf 'friend' entries.. standardized with
disallow=all, allow=gsm, and allow=ulaw
jitterbuffer=off
trunk=yes

I have explored a number of theories and none seem to be really helping the
situation.  The call drops are not consistent, so it is hard to say.

One thing I have considered is shared IRQs on some of the servers --- BUT
while I know this can affect TDM installations -- these machines have been
in production with no drops for months!

So does anyone have any suggestions as to why all of a sudden IAX calls and
my asterisk network would become so unstable?

Any suggestions appreciated

--
Chris Earle



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Re: [Asterisk-Users] asterisk-stat display problems

[Chris Earle \(CBL\)]

yep

I don't know exactly which things the php-gd is used for, but like I said,
someof the pages work, like the main record page, the little red bars
showing call volume work fine


Really annoying, cause it looks so good at that point, then you go to use
the other pages/features and it's broken

Thanks for the reply,

--
Chris



- Original Message - 
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Monday, June 26, 2006 1:49 PM
Subject: Re: [Asterisk-Users] asterisk-stat display problems


> do you have the php-gd package installed on your * server?
>
> Chris Earle (CBL) wrote:
> > Hey all,
> >
> > having a terrible time with asterisk-stat -- it runs, server is fine,
but
> > some of the pages don't display properly/at all --- I think this is a
code
> > problem with them, but not sure.  I thought everyone loved the
asterisk-stat
> > package?
> >
> > See below problems.  Any ideas?  Areski hasn't replied to me since
> >
> > --
> > Chris
> >
> >
> > - Original Message - 
> > From: "Chris Earle (CBL)"
> > To: "Areski"
> > Sent: Tuesday, June 13, 2006 6:15 PM
> > Subject: Re: CDR-Analyser version question
> >
> >
> >> Thank you for the reply;
> >>
> >> I see now that the main file cdr.php does work with argument ?s=1, 2,
> >> etc
> >> but when s=0, does not load
> >>
> >> I get an Apache error :
> >>
> >>  relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
> >> gdFontCacheShutdown
> >>
> >> Not sure if that means anything important;
> >>
> >>
> >>
> >>
> >> Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2),
the
> >> pages do not complete their output -- no search button displayed, stops
> >> outputting radio buttons for UserField row etc
> >>
> >> So at this point, only the main Call-log page (s=1) works.
> >>
> >>
> >> I am using Debian with php 4.4.1
> >> Mysql ver 12.22, Distrib 4.0.24
> >> GD Library is 2.0.33 I think
> >>
> >>
> >> Any input you can pass along would be much appreciated!  I am
comfortable
> >> with php so if you want me to modify sourcecode that is fine
> >>
> >> Thanks!
> >>
> >>
> >>
> >>
> >> - Original Message - 
> >> From: "Areski"
> >> To: "Chris Earle (CBL)"
> >> Sent: Sunday, May 28, 2006 7:11 PM
> >> Subject: Re: CDR-Analyser version question
> >>
> >>
> >>> No there is no asterisk requirement to make asterisk-stat.
> >>> Indeed the soft is only based on the cdr database. If you have an
error
> >>> you can give me more info, I may help you.
> >>>
> >>> Rgds, Areski
> >>>
> >>> On 5/25/06, Chris Earle (CBL) wrote:
> >>>> Hi there,
> >>>>
> >>>> quick question:
> >>>>
> >>>> Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using
Asterisk
> >> 1.0.x
> >>>> and can't get it to load the cdr.php properly
> >>>>
> >>>> so I downgraded to v1.3 and it works...
> >>>>
> >>>> Let me know if there's an asterisk version requirement for each
> > version
> >> of
> >>>> the CDR Analyser
> >>>>
> >>>> Thanks!
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Chris Earle
> >>>>
> >>>>
> > 
> >
> >
>
> -- 
> Mojo <[EMAIL PROTECTED]>
> Office Manger, Horan & Company, LLC
> (907) 747- x112
>
> -- 
> This message has been scanned for viruses and dangerous content by
> MailScanner, and is believed to be clean.


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[Asterisk-Users] asterisk-stat display problems

[Chris Earle \(CBL\)]

Hey all,

having a terrible time with asterisk-stat -- it runs, server is fine, but
some of the pages don't display properly/at all --- I think this is a code
problem with them, but not sure.  I thought everyone loved the asterisk-stat
package?

See below problems.  Any ideas?  Areski hasn't replied to me since

--
Chris


- Original Message - 
From: "Chris Earle (CBL)"
To: "Areski"
Sent: Tuesday, June 13, 2006 6:15 PM
Subject: Re: CDR-Analyser version question


> Thank you for the reply;
>
> I see now that the main file cdr.php does work with argument ?s=1, 2,
> etc
> but when s=0, does not load
>
> I get an Apache error :
>
>  relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol:
> gdFontCacheShutdown
>
> Not sure if that means anything important;
>
>
>
>
> Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the
> pages do not complete their output -- no search button displayed, stops
> outputting radio buttons for UserField row etc
>
> So at this point, only the main Call-log page (s=1) works.
>
>
> I am using Debian with php 4.4.1
> Mysql ver 12.22, Distrib 4.0.24
> GD Library is 2.0.33 I think
>
>
> Any input you can pass along would be much appreciated!  I am comfortable
> with php so if you want me to modify sourcecode that is fine
>
> Thanks!
>
>
>
>
> - Original Message - 
> From: "Areski"
> To: "Chris Earle (CBL)"
> Sent: Sunday, May 28, 2006 7:11 PM
> Subject: Re: CDR-Analyser version question
>
>
> > No there is no asterisk requirement to make asterisk-stat.
> > Indeed the soft is only based on the cdr database. If you have an error
> > you can give me more info, I may help you.
> >
> > Rgds, Areski
> >
> > On 5/25/06, Chris Earle (CBL) wrote:
> > > Hi there,
> > >
> > > quick question:
> > >
> > > Does asterisk-stat v2.0.1 require Asterisk 1.2+ ?  I am using Asterisk
> 1.0.x
> > > and can't get it to load the cdr.php properly
> > >
> > > so I downgraded to v1.3 and it works...
> > >
> > > Let me know if there's an asterisk version requirement for each
version
> of
> > > the CDR Analyser
> > >
> > > Thanks!
> > >
> > >
> > >
> > > --
> > > Chris Earle
> > >
> > >



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Re: [Asterisk-Users] ISDN in Japan?

[Chris Earle \(CBL\)]

So it could be BRI, PRI or maybe even Analog there??

I guess what I'm asking is it predominantly ISDN there or not

Thanks for the input about the card and chan-capi

:-)

--
Chris

- Original Message - 
From: "Armin Schindler" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Tuesday, April 18, 2006 1:32 PM
Subject: Re: [Asterisk-Users] ISDN in Japan?


> On Tue, 18 Apr 2006, Chris Earle (CBL) wrote:
> > Hi all,
> >
> > general query here --- I'm about to set up an asterisk box for use in
Japan
> > but can't figureout if it's all ISDN there or what?
> >
> > I have gathered so far that the two major providers, NTT and KVH both
offer
> > ISDN lines with ...INS1500 and maybe INS64 protocols?
> > Not sure...
> >
> > But I'm seeing stuff about J1 vs. T1/E1 
> > so does that mean I can't use a Digium card it there?
> >
> > Can someone please clarify what sort of system I'm looking at here and
if I
> > need a japanese retailer for the card or what
>
> I don't know the status of ISDN in Japan, but the Eicon DIVA Server cards
> (BRI and PRI) are provided with firmware for ISDN protocols in japan.
> Together with chan-capi it is fully functional with Asterisk/OpenPBX.
>
> Armin
>
> -- 
> This message has been scanned for viruses and dangerous content by
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[Asterisk-Users] ISDN in Japan?

[Chris Earle \(CBL\)]

Hi all,

general query here --- I'm about to set up an asterisk box for use in Japan
but can't figureout if it's all ISDN there or what?

I have gathered so far that the two major providers, NTT and KVH both offer
ISDN lines with ...INS1500 and maybe INS64 protocols?
Not sure...

But I'm seeing stuff about J1 vs. T1/E1 
so does that mean I can't use a Digium card it there?

Can someone please clarify what sort of system I'm looking at here and if I
need a japanese retailer for the card or what

;-)

Thanks!


--
Chris Earle
System Solutions Specialist


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[Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.

[Chris Earle]

Appreciate the replies everyone -- really

I'm wondering if I should be using zapHFC with my Junghanns card instead of
qozap?  Everyone always mentions zaphfc -- mostly I guessed because they are
using a zaphfc-compatible card - but *maybe* I should try that instead
of qozap???

And yep -- totally know about the module load order thing and ztcfg -- no
worries there

I've been able to dial out and everything from the start ! -- which is a
bridge from digium-->junghanns there..but incoming calls seem to be a
whole other issue. :-(


Exhausted from trying a million things,

Chris

"Chris Earle" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> What?  After hours of searching for anything to help me, I found this
> comment about zaptel cards in systems with bristuff-cards (junghanns for
me
> in this case)
>
> I havent' seen any other reports of this sort of behaviour --- can anyone
> confirm whether they've got a QuadBRI and TDM400P card working together in
> one machine?
>
>
> thanks :-S
>
>
>
> "Zoa" <[EMAIL PROTECTED]> wrote in message
> news:[EMAIL PROTECTED]
> >
> 
> >We stopped with the bristuff as bristuff will break any other zaptel
> >cards in the same system. (pri seems logical, why the tdm card also
> >broke is unknown to me).
> 
> > ___
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[Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.

[Chris Earle]

What?  After hours of searching for anything to help me, I found this
comment about zaptel cards in systems with bristuff-cards (junghanns for me
in this case)

I havent' seen any other reports of this sort of behaviour --- can anyone
confirm whether they've got a QuadBRI and TDM400P card working together in
one machine?


thanks :-S



"Zoa" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
>

>We stopped with the bristuff as bristuff will break any other zaptel
>cards in the same system. (pri seems logical, why the tdm card also
>broke is unknown to me).

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[Asterisk-Users] Re: Junghanns and Digium TDM400?

[Chris Earle]

OH ya, maybe some console output would clear this up for you:

Call comes in, goes into incoming call context...:

-- Executing NoOp("Zap/1-1", "49CALLERID") in new stack
-- Executing NoOp("Zap/1-1", "0") in new stack
-- Executing GotoIf("Zap/1-1", "0?s|4:s|5") in new stack
-- Goto (isdn-incoming,s,5)
-- Executing Dial("Zap/1-1", "IAX2/TESTPHONE/1001||Tt") in new stack
-- Called TESTPHONE/1001
-- Accepting data call from 'CALLERID' to 's' on channel 0/1, span 1
-- Call accepted by IAXPEER_SERVER_IP (format gsm)
-- Format for call is gsm
-- IAX2/TESTPHONE/16384 answered Zap/1-1
-- Hungup 'IAX2/TESTPHONE/16384'
  == Spawn extension (isdn-incoming, s, 5) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'


Asterisk (Bristuffed) 1.0.10, with Bristuff 0.2.0-RC8q (* 1.0.10)

I don't want to upgrade my asterisk and use the 0.3.0-PRE Bristuff
version. please tell me this will work another way



Any comments appreciated, anything!



"Chris Earle" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> This is getting very annoying.
>
> Thought it might be a irq conflict/sharing issue -- so resolved that.
>
> Still, cannot get an incoming call to bridge with IAX or ZAP devices.
>
> qozap: dropped audio..... errors remain as well
>
> Setup is in Germany -- TE mode,
>
> Someone plase help
> This is terrible.  Been trying everything for weeks
> :-(
>
>
> Chris
>
>
> "Chris Earle (CBL)" <[EMAIL PROTECTED]> wrote in message
> news:[EMAIL PROTECTED]
> > Hi all,
> >
> > is it possible to bridge a call between a Junghanns quadBRI card and a
> > TDM400 in the same server?
> >
> > It should be I think,  -- I am trying this and when an incoming call
comes
> > in, it hangs both up at the moment the bridge is attempted
> >
> > (and a subsequent 'qozap: dropped audio'  error is show in the
> > /var/log/messages)
> >
> >
> > Any thoughts appreciated -- I've seen posts, but no clear
> results/solutions
> >
> >
> >
> > --
> > Chris Earle
> > System Solutions Specialist
> >
> >
> > -- 
> > This message has been scanned for viruses and dangerous content by
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> >
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[Asterisk-Users] Re: Junghanns and Digium TDM400?

[Chris Earle]

This is getting very annoying.

Thought it might be a irq conflict/sharing issue -- so resolved that.

Still, cannot get an incoming call to bridge with IAX or ZAP devices.

qozap: dropped audio. errors remain as well

Setup is in Germany -- TE mode,

Someone plase help
This is terrible.  Been trying everything for weeks
:-(


Chris


"Chris Earle (CBL)" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Hi all,
>
> is it possible to bridge a call between a Junghanns quadBRI card and a
> TDM400 in the same server?
>
> It should be I think,  -- I am trying this and when an incoming call comes
> in, it hangs both up at the moment the bridge is attempted
>
> (and a subsequent 'qozap: dropped audio'  error is show in the
> /var/log/messages)
>
>
> Any thoughts appreciated -- I've seen posts, but no clear
results/solutions
>
>
>
> --
> Chris Earle
> System Solutions Specialist
>
>
> -- 
> This message has been scanned for viruses and dangerous content by
> MailScanner, and is believed to be clean.
>
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Re: [Asterisk-Users] pickup problem

[Chris Earle \(CBL\)]

Ha -- this looks useful

Just was trying to do a *8 on an IAXy phone...realized it didn't work
across protocols

If I implement this, I'll have to code in *8 into my extensions.conf instead
of relying on the default built in 'steal' ?

--
Chris


- Original Message - 
From: "Mimmus" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
; <[EMAIL PROTECTED]>
Sent: Monday, March 20, 2006 1:17 PM
Subject: RE: [Asterisk-Users] pickup problem


> PickUp2:
>  http://linux.thorsten-knabe.de/asterisk/pickup.jsp
> works very well.
>
> Mimmus
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Tim Panton
> > Sent: Monday, March 20, 2006 4:50 PM
> > To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> > Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] pickup problem
> >
> >
> > On 20 Mar 2006, at 15:39, Rich Adamson wrote:
> >
> > > Mimmus wrote:
> > >>> -Original Message-
> > >>> From: [EMAIL PROTECTED] [mailto:asterisk-
> > >>> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> > >>> Sent: Monday, March 20, 2006 4:06 PM
> > >>>
> > >>> there is also a more generic call pickup using 'callgroup=2' and
> > >>> 'pickupgroup=2' in your sip definitions. That approach uses *8 or
> > >>> *8# to pickup any ringing phone within the callgroup number (eg,
> > >>> 2 in this example).
> > >> Does this call pickup work with IAX2?
> > >> If yes, how, if there is no callgroup/pickupgroup setting in
> > >> iax.conf?
> > >> More in general: does call pickup work between different protocols?
> > >
> > > Never had a need to do pickup with iax, so don't have a clue.
> > >
> > > As I recall, the callgroup keyword only applies to sip and zap
> > > channels.
> >
> > It doesn't work between protocols.
> >
> >
> > Tim Panton
>
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[Asterisk-Users] Junghanns and Digium TDM400?

[Chris Earle \(CBL\)]

Hi all,

is it possible to bridge a call between a Junghanns quadBRI card and a
TDM400 in the same server?

It should be I think,  -- I am trying this and when an incoming call comes
in, it hangs both up at the moment the bridge is attempted

(and a subsequent 'qozap: dropped audio'  error is show in the
/var/log/messages)


Any thoughts appreciated -- I've seen posts, but no clear results/solutions



--
Chris Earle
System Solutions Specialist


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[Asterisk-Users] qozap drops -- possible to bridge BRIstuff ISDN to analog zaptel phone?

[Chris Earle \(CBL\)]

Hi all,

Using Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q
with a Junghanns quadBRI (2 spans connected) and a Digium TDM400 for
extensions

Shouuld I be worried about these lines that keep showing up in my
/var/log/messages?

qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116
qozap: dropped audio card 1 cardid 0 bytes 6 z1 50 z2 28
qozap: dropped audio card 1 cardid 0 bytes 6 z1 59 z2 37
..



This may or may not be related to a problem I am having with incoming calls.

They seem to appear when my incoming calls keep getting dropped as soon as
you pick up the analog handset to bridge.
Call comes in, rings analog handsetsbut then everything hangs up as
soon as one of the dialed handsets is picked up.


Stumped on this ISDN dialplan stuff...new to me


I've seen mention on voip-info about an edit to the qozap source -- 
http://www.voip-info.org/wiki/index.php?page=zaptelBRI
But that doesn't seem to be my exact problem, as system is loading, asterisk
runs..can dial out too!

suggestions appreciated!


--
Chris Earle
System Solutions Specialist

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Re: [Asterisk-Users] module load order for Junghanns qozap and TDMcard

[Chris Earle \(CBL\)]

Okay

think I finally figured this out

it's the modules.conf post-install lines that run ztcfg

You're not supposed to run ztcfg more than once with the multiple zaptel
cards in there  I kept running it manually (ztcfg -) not realizing
that after modprobe wcfxs the ztcfg was being run.

So the order that works is

zaptel
qozap
wcfxs (which runs ztcfg, and readies asterisk to run)


If anyone has any comments about this, please post


--
Chris


- Original Message - 
From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Thursday, March 16, 2006 11:30 AM
Subject: Re: [Asterisk-Users] module load order for Junghanns qozap and
TDMcard


> Maybe this will shed some light about what I'm trying to do:
>
> This is some output from dmesg after this load order:
>
> modprobe zaptel
> insmod wcfxs
> insmod qozap
> 
> Zapata Telephony Interface Registered on major 196
> Freshmaker version: 73
> Freshmaker passed register test
> Module 0: Installed -- AUTO FXS/DPO
> Module 1: Installed -- AUTO FXS/DPO
> Module 2: Installed -- AUTO FXS/DPO
> Module 3: Installed -- AUTO FXS/DPO
> Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
> PCI: Enabling device 02:01.0 ( -> 0003)
> qozap: Junghanns.NET quadBRI card configured at mem 0xf889b000 IRQ 17 HZ
100
> CardID 0
> qozap: S/T ports: 4 [ TE TE TE TE ]
> qozap: 1 multiBRI card(s) in this box, 4 BRI ports total.
>
>
> Zaptel Configuration
> ==
>
> SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
> SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
> SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
> SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1)
>
> Channel map:
>
> Channel 01: Individual Clear channel (Default) (Slaves: 01)
> Channel 02: Individual Clear channel (Default) (Slaves: 02)
> Channel 03: D-channel (Default) (Slaves: 03)
> Channel 04: Individual Clear channel (Default) (Slaves: 04)
> Channel 05: Individual Clear channel (Default) (Slaves: 05)
> Channel 06: D-channel (Default) (Slaves: 06)
> Channel 07: Individual Clear channel (Default) (Slaves: 07)
> Channel 08: Individual Clear channel (Default) (Slaves: 08)
> Channel 09: D-channel (Default) (Slaves: 09)
> Channel 10: Individual Clear channel (Default) (Slaves: 10)
> Channel 11: Individual Clear channel (Default) (Slaves: 11)
> Channel 12: D-channel (Default) (Slaves: 12)
> Channel 13: FXO Kewlstart (Default) (Slaves: 13)
> Channel 14: FXO Kewlstart (Default) (Slaves: 14)
> Channel 15: FXO Kewlstart (Default) (Slaves: 15)
> Channel 16: FXO Kewlstart (Default) (Slaves: 16)
>
> 16 channels configured.
>
> ZT_SPANCONFIG failed on span 1: Invalid argument (22)
>
>
>
>
> any thoughts?
>
>
>
>
> Chris
>
>
>
>
> - Original Message - 
> From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Sent: Thursday, March 16, 2006 10:09 AM
> Subject: [Asterisk-Users] module load order for Junghanns qozap and TDM
card
>
>
> > Hi all,
> >
> > I'm trying to get a junghanns QuadBRI to coexist in the same machine as
a
> > Digium TDM400P card  (so I can run the ISDN lines in and bridge with
> analog
> > phones plugged into the TDM).
> >
> > I'm having a problem loading the modules.  If I follow the BRIstuff
> > (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod
> > qozap.o
> > I'm on Debian 2.4.31.
> > That works.
> > But then I still need the Digium module. (modprobe wctdm)
> > I've tried a few different orders.  Sometimes I can get the digium to
> load,
> > and the qozap.
> > but then I get an error on the ztcfg about Span  invalid argument (could
> be
> > my zaptel.conf I realize...)
> >
> > *If* I try loading the wctdm after the zaptel and qozap, the server
> freezes!
> > Some loop about qozap - dropped audio card
> >
> > I don't know if the quadBRI and the TDM are conflicting/sharing the
zaptel
> > module, or if I need to modprobe zaptel before each of them? and in what
> > order?
> >
> > Any suggestions appreciated... I haven't even got to figuring out what I
> can
> > do with chan_capi, just want to get the BRI card on and stuff.
> >
> > Thanks for any ideas!
> >
> >
> > --
> > Chris Earle


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Re: [Asterisk-Users] module load order for Junghanns qozap and TDM card

[Chris Earle \(CBL\)]

Maybe this will shed some light about what I'm trying to do:

This is some output from dmesg after this load order:

modprobe zaptel
insmod wcfxs
insmod qozap

Zapata Telephony Interface Registered on major 196
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXS/DPO
Module 3: Installed -- AUTO FXS/DPO
Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
PCI: Enabling device 02:01.0 ( -> 0003)
qozap: Junghanns.NET quadBRI card configured at mem 0xf889b000 IRQ 17 HZ 100
CardID 0
qozap: S/T ports: 4 [ TE TE TE TE ]
qozap: 1 multiBRI card(s) in this box, 4 BRI ports total.


Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: D-channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: D-channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: D-channel (Default) (Slaves: 12)
Channel 13: FXO Kewlstart (Default) (Slaves: 13)
Channel 14: FXO Kewlstart (Default) (Slaves: 14)
Channel 15: FXO Kewlstart (Default) (Slaves: 15)
Channel 16: FXO Kewlstart (Default) (Slaves: 16)

16 channels configured.

ZT_SPANCONFIG failed on span 1: Invalid argument (22)




any thoughts?




Chris




- Original Message - 
From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Sent: Thursday, March 16, 2006 10:09 AM
Subject: [Asterisk-Users] module load order for Junghanns qozap and TDM card


> Hi all,
>
> I'm trying to get a junghanns QuadBRI to coexist in the same machine as a
> Digium TDM400P card  (so I can run the ISDN lines in and bridge with
analog
> phones plugged into the TDM).
>
> I'm having a problem loading the modules.  If I follow the BRIstuff
> (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod
> qozap.o
> I'm on Debian 2.4.31.
> That works.
> But then I still need the Digium module. (modprobe wctdm)
> I've tried a few different orders.  Sometimes I can get the digium to
load,
> and the qozap.
> but then I get an error on the ztcfg about Span  invalid argument (could
be
> my zaptel.conf I realize...)
>
> *If* I try loading the wctdm after the zaptel and qozap, the server
freezes!
> Some loop about qozap - dropped audio card
>
> I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel
> module, or if I need to modprobe zaptel before each of them? and in what
> order?
>
> Any suggestions appreciated... I haven't even got to figuring out what I
can
> do with chan_capi, just want to get the BRI card on and stuff.
>
> Thanks for any ideas!
>
>
> --
> Chris Earle
>
>
> -- 
> This message has been scanned for viruses and dangerous content by
> MailScanner, and is believed to be clean.
>
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[Asterisk-Users] module load order for Junghanns qozap and TDM card

[Chris Earle \(CBL\)]

Hi all,

I'm trying to get a junghanns QuadBRI to coexist in the same machine as a
Digium TDM400P card  (so I can run the ISDN lines in and bridge with analog
phones plugged into the TDM).

I'm having a problem loading the modules.  If I follow the BRIstuff
(0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod
qozap.o
I'm on Debian 2.4.31.
That works.
But then I still need the Digium module. (modprobe wctdm)
I've tried a few different orders.  Sometimes I can get the digium to load,
and the qozap.
but then I get an error on the ztcfg about Span  invalid argument (could be
my zaptel.conf I realize...)

*If* I try loading the wctdm after the zaptel and qozap, the server freezes!
Some loop about qozap - dropped audio card

I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel
module, or if I need to modprobe zaptel before each of them? and in what
order?

Any suggestions appreciated... I haven't even got to figuring out what I can
do with chan_capi, just want to get the BRI card on and stuff.

Thanks for any ideas!


--
Chris Earle


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Re: [Asterisk-Users] Junghanns, Germany ISDN settings

[Chris Earle \(CBL\)]

Thanks for the info, I am confused still ;-)

It sounds like I need NT mode -- there are NTBA boxes involved at my
location...

And then -- what do I do about

Termination of S/T Interface ??

and

Power Feeding?

http://www.junghanns.net/downloads/quadbrijumpersnew.pdf

That's what I'm referencing

Someone feed me some tips please!



- Original Message - 
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Saturday, March 11, 2006 5:08 AM
Subject: Re: [Asterisk-Users] Junghanns, Germany ISDN settings


> Hi Chris,
>
> Chris Earle (CBL) wrote:
> > I've got a Junghanns QuadBRI card which I'm going to install on a system
in
> > Germany
> >
> > Anyone give me some tips on the Jumper settings?  I'm guessing it's
going to
> > be NT mode with p2p?  I haven't used ISDN before.
> >
> > I'm going to also put a Digium TDM400P card in there to plug the analog
> > phones into.
> >
> > I'm just worried about the jumpers and modes.
>
> It really depends what you will be hooking up to the asterisk box. If
> you are connecting to a telco's S0 bus you want the card to be in TE
> mode (Terminal Equipment). If you are using multiple ISDN lines that are
> coupled together as one bundle (ask the telco) you will probably neet to
> configure it as p2p. If all lines are singular, use p2mp.
>
> If you will be connecting to a PBX, everything is dependant on how that
> PBX is configured.
>
>
> Best regards,
> Florian
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[Asterisk-Users] Junghanns, Germany ISDN settings

[Chris Earle \(CBL\)]

Hi all,

I've got a Junghanns QuadBRI card which I'm going to install on a system in
Germany

Anyone give me some tips on the Jumper settings?  I'm guessing it's going to
be NT mode with p2p?  I haven't used ISDN before.

I'm going to also put a Digium TDM400P card in there to plug the analog
phones into.

I'm just worried about the jumpers and modes.


Suggestions appreciated,

--
Chris Earle
System Solutions Specialist



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Re: [Asterisk-Users] Auto login via Remote User

[Chris Earle \(CBL\)]

IGNORE, mispost!

:-S


- Original Message - 
From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, February 28, 2006 4:59 PM
Subject: [Asterisk-Users] Auto login via Remote User


> This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER
> requires web server  authentication right?
>
> Correct me if I'm wrong, but this means that the contents of the User
Table
> would have to be in the passwd file defined through the .htaccess file
> right?
>
> .. because it passes whatever the user authenticates on the webserver with
> through to the wiki/extensions scripts right?
>
> (resulting in this not really being useful because i've got to duplicate
my
> user table )
>
>
> Comments appreciated,
>
> --
> Chris Earle
> System Solutions Specialist
>
>
> -- 
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[Asterisk-Users] Auto login via Remote User

[Chris Earle \(CBL\)]

This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER
requires web server  authentication right?

Correct me if I'm wrong, but this means that the contents of the User Table
would have to be in the passwd file defined through the .htaccess file
right?

.. because it passes whatever the user authenticates on the webserver with
through to the wiki/extensions scripts right?

(resulting in this not really being useful because i've got to duplicate my
user table )


Comments appreciated,

--
Chris Earle
System Solutions Specialist


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Re: [Asterisk-Users] Cannot boot machine up after working on zaptel....

[Chris Earle \(CBL\)]

Thanks

Yeah, you would think so wouldn't you.

Tried that , and still wouldn't boot

Really annoying. beacuse I've been doing work with the zaptel drivers
and such and this happened once already...

Thanks for the suggestion,

Chris

- Original Message - 
From: "Colin Anderson" <[EMAIL PROTECTED]>
To: "'Chris Earle (CBL)'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing
List - Non-Commercial Discussion'" 
Sent: Tuesday, February 28, 2006 10:05 AM
Subject: RE: [Asterisk-Users] Cannot boot machine up after working on
zaptel


> What happens if you take out the Zaptel I/F's? If it boots, you can
correct
> whatever you did then replace them.
>
> hth
>
> -Original Message-
> From: Chris Earle (CBL) [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, February 28, 2006 7:45 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Cannot boot machine up after working on
> zaptel
>
>
> Hi all,
>
> hard for me to explain this, but it keeps happening on a number of
machines
>
> I attempt to upgrade zaptel, or do something to zaptel modules. and
then
> I reboot the machine, and for whatever reason, it hangs on loading the
> modules
>
> Either the install wasn't complete, the zaptel modules settings are wrong,
> whatever
> but the problem is now I can't get past the boot up and the machine is
> basically lost
>
> Is there any way to bypass the module load attempt or anything?
>
> I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD,
> but no go
>
> I'm on Debian 2.4.18, with Zaptel 1.0.9.2
>
> I understand that there was something wrong in the modules config, but
> surely I should be able to bypass and get back in to fix it!
>
> Any ideas greatly appreciated, as I would rather not have to use an old
> clone drive and start over
>
>
> --
> Chris Earle
> System Solutions Specialist
>
>
> -- 
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[Asterisk-Users] Cannot boot machine up after working on zaptel....

[Chris Earle \(CBL\)]

Hi all,

hard for me to explain this, but it keeps happening on a number of machines

I attempt to upgrade zaptel, or do something to zaptel modules. and then
I reboot the machine, and for whatever reason, it hangs on loading the
modules

Either the install wasn't complete, the zaptel modules settings are wrong,
whatever
but the problem is now I can't get past the boot up and the machine is
basically lost

Is there any way to bypass the module load attempt or anything?

I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD,
but no go

I'm on Debian 2.4.18, with Zaptel 1.0.9.2

I understand that there was something wrong in the modules config, but
surely I should be able to bypass and get back in to fix it!

Any ideas greatly appreciated, as I would rather not have to use an old
clone drive and start over


--
Chris Earle
System Solutions Specialist


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Re: [Asterisk-Users] indications issues in Singapore?

[Chris Earle \(CBL\)]

Good to know about that Loopstart thing --- helped me quickly solve my
problem of the phones not ringing :-)

thank you for the input


Chris


- Original Message - 
From: "Leo Ann Boon" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Friday, February 17, 2006 7:10 PM
Subject: Re: [Asterisk-Users] indications issues in Singapore?


> Chris Earle (CBL) wrote:
>
> >Hi all,
> >
> >haven't seen many posts about asterisk in Singapore...
> >Getting a server going there and was wondering if TDM400Ps will be fine
in
> >FCC mode, and if there are indications / cadence values that I should be
> >putting on there as in other international locations.
> >
> >Seen an unsettling post on voip-info about Singapore issues with Call
> >Polarity/Hangup Detection -- crossing my fingers I don't run into that
> >problem :-)
> >
> >
> Analog lines here are mostly loopstart, so you need to enable busydetect
> if you're using the zaptel FXO. A better option is to use a capi ISDN
> BRI card. I used the Fritz! PCI card with chan_capi, costs around S$160.
> The original poster on voip-info wrote about using kewlstart and CPC,
> which I have never encountered over here. I guess it was in vogue during
> the good old DID analog trunk days. But nowadays, you either use plain
> analog or move to BRI/PRI if you need MSN/DDI.
>
>
>
>
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[Asterisk-Users] indications issues in Singapore?

[Chris Earle \(CBL\)]

Hi all,

haven't seen many posts about asterisk in Singapore...
Getting a server going there and was wondering if TDM400Ps will be fine in
FCC mode, and if there are indications / cadence values that I should be
putting on there as in other international locations.

Seen an unsettling post on voip-info about Singapore issues with Call
Polarity/Hangup Detection -- crossing my fingers I don't run into that
problem :-)

Any tips appreciated,


--
Chris Earle
System Solutions Specialist


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[Asterisk-Users] Issues in Australia? Ringing, iaxy etc

[Chris Earle \(CBL\)]

Hi all,

getting a server going wiht a few TDM400's and some phones, and some IAXys
too

I haven't heard any issues about AU phones being able to RING in Australia,
like the problem in the UK with ring capacitors on the BT system.  Are there
any problems like that?

Also, with the iaxy's -- they should work (and ring) in Australia right?
The only hint I'm seeing around is the use of notransfer=yes in the iax.conf
for the iaxy entry

Basically, just hoping for a smooth transition over to the asterisk
system....

Cheers


--
Chris Earle
System Solutions Specialist


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Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN

[Chris Earle \(CBL\)]

Hey all

would you believe this problem is still going on?

I have now successfully secured individual IRQs for the two TDM cards, but
no change has been observed.  No interrupts appear to be getting dropped or
anything along those lines.

Also tried moving the FXO and FXS modules around on the cards.  No
changes -- not completely clear if the card is defective or not.  Can you
shed any light on that?

I am using Zaptel 1.2.2, with the MG2 echo canceller, but only with Asterisk
1.0.9.  I am not prepared to upgrade to a more recent 1.2 version unless I
think that will provide massive benefits (I don't think it will affect my
zaptel/echo problem?)

Back to the card situation: we have not tried any other machines or cards.
We are starting to consider using a different card -- and would prefer to
stick with Digium hardware.  The newer, larger, TDM2400 is under
consideration because it has echo cancellation onboard --- Do you think this
will do anything??

Also, tried the phone on the phoneline straight up -- sounded fine.  We
tried the phoneline with a US phone and also a UK
one.

So is hardware EC the solution to my woes?



Chris

- Original Message - 
From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
To: "Chris Bagnall" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List -
Non-Commercial Discussion'" 
Sent: Monday, January 23, 2006 5:24 PM
Subject: Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN


> Thanks for that, insightful
>
> The quality on both ends with those settings is quite good -- but the
> extension side still has a ridiculous -- almost duplicate -- echo!
>
> If I turn the txgain right down, I lose all sound.of course the
signals
> in ztmonitor show up perfect then, but can't hear anything (DTMF tones too
> low or whatever...)
>
> Still wondering if it's an impedance issue...or something along those
> lines/chipset..etc
>
> I'm going to attempt upgrading Zaptel now, without upgrading the asterisk
>
>
> Chris
>
>
> - Original Message - 
> From: "Chris Bagnall" <[EMAIL PROTECTED]>
> To: "'Chris Earle (CBL)'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing
> List - Non-Commercial Discussion'" 
> Sent: Sunday, January 22, 2006 9:13 AM
> Subject: RE: [Asterisk-Users] Re: Connection TDM400P to UK PSTN
>
>
> > > Successfully got the adapters to allow the BT phones to ring
> > > on lines coming out of a TDM.. but now my latest
> > > problem is echo.
> > > Suggestions / Experiences in UK appreciated
> >
> > Most of our clients with BT lines tend to have ISDN BRIs, but we do have
> one
> > in Northampton running 3 analogue lines from a TDM400.
> >
> > zaptel.conf is as follows:
> > fxsks   = 1-3
> > loadzone= uk
> > defaultzone = uk
> >
> > The TDM driver is loaded with opermode=UK and the output from dmesg
> confirms
> > this.
> >
> > Relevant settings from zapata.conf are as follows:
> > echocancel=yes
> > echocancelwhenbridged=yes
> > echotraining=800
> > rxgain=8.0
> > txgain=-4.0
> > busydetect=yes
> > group   = 1
> > context = inbound
> > channel => 1-2
> > group   = 2
> > context = inbound
> > channel => 3
> >
> > They're running Asterisk/Zaptel 1.0.10. There were major echo issues
when
> we
> > first deployed the system back in September, but some careful tweaking
of
> > rxgain and txgain seems to have largely resolved the situation.
Certainly
> my
> > experience has been that rxgain and txgain have far more impact on echo
> > reduction than any of the echo-specific settings. Get the gains right
> first,
> > then play with the echo-specific settings.
> >
> > Regards,


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Re: [Asterisk-Users] Fritz card technology & German *

[Chris Earle \(CBL\)]

Hi all --
I just sat down to revisit this project and figure out what card to go with.

Something that no one around me seemed to consider was what Legacy PBX is in
place already.  There is a PBX with analog phones going into it!

So if I get a 4x BRI card, I'm going to either need to keep the Legacy PBX
in place and *somehow* connect to the two machines, or I'm going to need a
TDM card in the server as well and handle everything inside the one server
with Asterisk.

This is driving me up the wall, because I'm not physically in the same
location as the German PBX/where the Asterisk server is going to be setup.
I am also not clear on how ISDN works with incoming and outgoing lines.
Does it distinguish between the incoming lines/outgoing?  Meaning -- will
plugging 2x CAT5 lines between the BRIcard/Asterisk and the Legacy PBX to
enable bridging between the two??

Any further comments on this would be appreciated


Man is ISDN wacky :-)   I'd much rather just get a large TDM2400 and put all
the extension lines into that.

Cheers all

- Original Message - 
From: "Kristof Hardy" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, January 19, 2006 11:59 AM
Subject: Re: [Asterisk-Users] Fritz card technology & German *


> Chris Earle (CBL) wrote:
> > I have also heard about BERONET isdn cards?  a single Beronet 4-channel
card
> > would suffice I think?
>
> Yes. Beronet and Junghanns both have the "same" cards. (they just 'work'
> different, junghanns uses zap interfaces, beronet mISDN)
>
> So, as already mentioned, you have 2 good options:
> - 4x BRI card (Beronet or junghanns)
> - 2x HFC PCI card (uses zap, and are cheap!)
>
> Regarding the phones, I only use sip phones, so no idea on that..
>
>
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Re: SV: [Asterisk-Users] BAD/GOOD Echo Cancel

[Chris Earle \(CBL\)]

Okay -- I've got serious echo problem with my BT line in the UK, and a
TDM400P

So I'm starting to consider getting the TDM2400 instead, to take advantage
of its hardware echo cancellation -- is this a reasonable idea?  I'm all
analog, so the TExxx cards aren't an option


Comments appreciated,

Chris Earle
System Solutions Specialist

- Original Message - 
From: "Jerry Jones" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, February 06, 2006 8:19 AM
Subject: Re: SV: [Asterisk-Users] BAD/GOOD Echo Cancel


>
> On Feb 6, 2006, at 5:04 AM, <[EMAIL PROTECTED]>
> <[EMAIL PROTECTED]> wrote:
>
> > Im curious. Does anyone have experienced echo-problems that later
> > where solved by buying a hardware-echo canceller such as the
> > Wildcard TE411P?
>
> Yes. I turned off all echo can on the wildcards and bought external.
> Point towards carrier and works like a charm. btw - use acoustic echo
> can not hybrid ec.
>
>
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Re: [Asterisk-Users] S100-FX v2.0

[Chris Earle \(CBL\)]

Curious about this device as well.  Seems 
almost too good to be true?
The built-in switch feature would be much handier 
than having to get a router for the extra network connection;
 
Also, the 'life line passthru' thing seems 
interesting -- although I have no idea what a life line passthru is! haha  
It would be amazing if it was an FXO, but obviously isn't -- I'm assuming you 
can dial out on that line if necessary? 'life line' concept?
 
 
Anyone had a good experience with one of these IN 
THE UK?? (or any country outside of North America for that matter)  power 
issues etc
 
 
Regards
 

  - Original Message - 
  From: 
  Mike Hammett 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, January 27, 2006 1:47 
  AM
  Subject: [Asterisk-Users] S100-FX 
  v2.0
  
  I just saw the S100-FX v2.0 on eBay.  I was wondering if anyone has tried it out 
  and what their opinion of it was.
   
   
  Mike HammettIntelligent Computing 
  Solutionshttp://www.ics-il.com
   
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Re: [Asterisk-Users] Need to recompile * after changing zap echo method?

[Chris Earle \(CBL\)]

Hi there,

Don't think you have to recompile * if you've already compiled * with zaptel
before.  (chan_zap.so exists)

Should just have to rebuild zaptel, install the module, and do a ztcfg

Good luck,



- Original Message - 
From: "Brent Torrenga" <[EMAIL PROTECTED]>
To: 
Sent: Monday, January 30, 2006 11:34 AM
Subject: [Asterisk-Users] Need to recompile * after changing zap echo
method?


> Dearest List,
>
> I guess I missed this point: Is it true that if you change the echo
canceler
> in zconfig.h, and then recompile/install your zap modules, that for this
to
> be taken into effect by * you must then recompile/install *?
>
> I would have figured that the zap echo cancellation method was independent
> of *, and I don't recall seeing any docs mentioning either way.
>
>
> Sincerely,
>
> Brent A. Torrenga
> [EMAIL PROTECTED]
>
> Torrenga Engineering, Inc.
> 907 Ridge Road
> Munster, Indiana 46321-1771
>
> 219.836.8918x325 Voice
> 219.836.1138 Facsimile
> www.torrenga.com
>
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Re: [Asterisk-Users] Re: Point to Point with Fritz Card ...

[Chris Earle \(CBL\)]

I am venturing down this path soon, trying to 
decide if I will be okay sticking with 2 x Fritz! PCI cards or if I should 
really get a different card... 4 port ISDN or something...
Chan_Capi with 2 Fritz cards I'm told requires code 
modifications to work etc...?
 
Suggestions?
 
 
Chris Earle,
System Solutions Specialist

  - Original Message - 
  From: 
  Dias Badekas 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, September 20, 2005 7:23 
  AM
  Subject: [Asterisk-Users] Re: Point to 
  Point with Fritz Card ...
  I 
  just set up a system with two ISDN pci cards and am using mISDN, plus 
  chan_misdn (multipoint only).It seems to work fine except for a few 
  annoyances, as I wrote in another post.I tried to ran chan_capi, 
  afterwords, just to check on the difference but had problems. Of course, I 
  did not load chan_misdn and chan_capi together as they are mutually exclusive, 
  as per docs.Have you been successful in running chan_capi using the mISDN 
  drivers? The misdn docs say you should be able, but after trying once I 
  wouldlike to hear experiences on this. Chan_capi has a lot of features 
  plus fax stuff implemented that make it interesting.DBOn 
  Tue, 2005-09-20 at 13:15 +0800, Craig Guy wrote: 
  You will need to use the mISDN drivers - the AVM CAPI drivers will not 
support PTP.  It is possible to use mISDN with chan_capi but chan_misdn 
would be easier.

Craig

- Original Message - 
From: "Joao Correia" <[EMAIL PROTECTED]>
To: <asterisk-users@lists.digium.com>
Sent: Tuesday, September 20, 2005 4:57 AM
Subject: [Asterisk-Users] Point to Point with Fritz Card ...


> Hello all,
>
> Does anyone has any experience with Point to Point Fritz Card and 
> Asterisk ?
>
> I have a BRI access Point to Multipoint working fine but I can only  have 
> 3 numbers.
>
> The phone telco said that if they change to Point to Point I can have  10 
> numbers.
>
> Does anyone has any experience with Point to Point ?
>
> Best regards
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Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN

[Chris Earle \(CBL\)]

Thanks for that, insightful

The quality on both ends with those settings is quite good -- but the
extension side still has a ridiculous -- almost duplicate -- echo!

If I turn the txgain right down, I lose all sound.of course the signals
in ztmonitor show up perfect then, but can't hear anything (DTMF tones too
low or whatever...)

Still wondering if it's an impedance issue...or something along those
lines/chipset..etc

I'm going to attempt upgrading Zaptel now, without upgrading the asterisk


Chris


- Original Message - 
From: "Chris Bagnall" <[EMAIL PROTECTED]>
To: "'Chris Earle (CBL)'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing
List - Non-Commercial Discussion'" 
Sent: Sunday, January 22, 2006 9:13 AM
Subject: RE: [Asterisk-Users] Re: Connection TDM400P to UK PSTN


> > Successfully got the adapters to allow the BT phones to ring
> > on lines coming out of a TDM.. but now my latest
> > problem is echo.
> > Suggestions / Experiences in UK appreciated
>
> Most of our clients with BT lines tend to have ISDN BRIs, but we do have
one
> in Northampton running 3 analogue lines from a TDM400.
>
> zaptel.conf is as follows:
> fxsks   = 1-3
> loadzone= uk
> defaultzone = uk
>
> The TDM driver is loaded with opermode=UK and the output from dmesg
confirms
> this.
>
> Relevant settings from zapata.conf are as follows:
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=800
> rxgain=8.0
> txgain=-4.0
> busydetect=yes
> group   = 1
> context = inbound
> channel => 1-2
> group   = 2
> context = inbound
> channel => 3
>
> They're running Asterisk/Zaptel 1.0.10. There were major echo issues when
we
> first deployed the system back in September, but some careful tweaking of
> rxgain and txgain seems to have largely resolved the situation. Certainly
my
> experience has been that rxgain and txgain have far more impact on echo
> reduction than any of the echo-specific settings. Get the gains right
first,
> then play with the echo-specific settings.
>
> Regards,
>
> Chris
> -- 
> C.M. Bagnall, Director, Minotaur I.T. Limited
> This email is made from 100% recycled electrons
>
>
>
> -- 
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Re: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of *

[Chris Earle \(CBL\)]

Has upgrading to the newer Zaptel allowed you to use the newer improvements
in it? (sorry if that was implied)

Thanks for the speedy reply


Chris


- Original Message - 
From: "Adam Robins" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Monday, January 23, 2006 5:01 PM
Subject: RE: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of *


> I have done this successfully with Asterisk 1.07 and Zaptel 1.09 and
> 1.2.1 for the same reasons as you.
>
> However, if you ever need to go recompile Asterisk, then you will first
> need to recompile the old Zaptel, compile Asterisk and the new Zaptel
> again.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Chris
> Earle (CBL)
> Sent: Monday, January 23, 2006 4:41 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of *
>
> Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2
> whatever) with an older version of Asterisk? I'm running 1.09, but I was
> wondering if I could get at the newer echo cancellers like KB1 and MG2
> without upgrading to Asterisk 1.2?
>
>
> I'm going out on a limb here to try and fix a serious echo problem on a
> TDM
> + BT PSTN line in the UK
>
>
> Thanks for your suggestions everyone
>
>
> --
> Chris Earle
> System Solutions Specialist,
>
>
> --
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[Asterisk-Users] Newer version of Zaptel with 1.0 branch of *

[Chris Earle \(CBL\)]

Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever)
with an older version of Asterisk? I'm running 1.09, but I was wondering if
I could get at the newer echo cancellers like KB1 and MG2 without upgrading
to Asterisk 1.2?


I'm going out on a limb here to try and fix a serious echo problem on a TDM
+ BT PSTN line in the UK


Thanks for your suggestions everyone


--
Chris Earle
System Solutions Specialist,


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[Asterisk-Users] Connection TDM400P to UK PSTN (Gains and Impedance)

[Chris Earle \(CBL\)]

re: setting zaptel to UK mode
Yes, I've set the tonezone, and also when loading the module have set it to
OPERMODE=UK
Looks like that is working

A few ideas:  *Maybe* the opermode code in the zaptel drivers isn't actually
working ? Meaning, maybe it's saying it's in UK mode but actually isn't?  Is
there any way to know really?

Another question:
Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever)
with an older version of Asterisk? I'm running 1.09, but I was wondering if
I could get at the newer echo cancellers like KB1 and MG2 without upgrading
to Asterisk 1.2?

Thanks for your suggestions everyone

Chris


- Original Message - 
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
Sent: Thursday, January 19, 2006 1:02 PM
Subject: Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN


> On Thu, Jan 19, 2006 at 11:30:25AM -0500, Chris Earle (CBL) wrote:
> > Okay, sorry to hash out this discussion again, but it's starting to
drive me
> > crazy
> >
> > Successfully got the adapters to allow the BT phones to ring on lines
coming
> > out of a TDM.. but now my latest problem is echo.
> >
> > I have done tweaking of the gains in North and South America, and after
a
> > bit of work have gotten echo to go away, but this seems to just not want
to
> > go away.
> >
> > On an incoming call from the POTS, everything on my end sounds perfect,
> > but on the internal extension phone, there is an echo when you speak.
An
> > almost perfect copy of what you say.  If I turn down the gains on that
> > channel, it doesn't seem to do much, or causes other volume issues.
> >
> > Help!
> >
> > In my research and hunting, I am starting to worry that the US-bought
digium
> > cards have IMPEDENCE issues in the UK with the BT Lines etc?  That would
> > seem to explain why the echo is so incessant.  I have even tried
changing
> > Echo Cancellers to MARK3.
> >
> > Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian
> >
> > Suggestions / Experiences in UK appreciated
>
> Not sure, but if the card is recent, then its impedence values and such
> should be configurable.
>
> For starters, set the tonoezone (both of them) to "uk" in zaptel.conf
> and re-run ztcfg .
>
> Try to see what exactly is the model of your card and your firmware. I'm
> not sure how to check this. Maybe ask Digium support.
>
> -- 
> Tzafrir Cohen | [EMAIL PROTECTED] | VIM is


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[Asterisk-Users] Re: Connection TDM400P to UK PSTN

[Chris Earle \(CBL\)]

Okay, sorry to hash out this discussion again, but it's starting to drive me
crazy

Successfully got the adapters to allow the BT phones to ring on lines coming
out of a TDM.. but now my latest problem is echo.

I have done tweaking of the gains in North and South America, and after a
bit of work have gotten echo to go away, but this seems to just not want to
go away.

On an incoming call from the POTS, everything on my end sounds perfect,
but on the internal extension phone, there is an echo when you speak.  An
almost perfect copy of what you say.  If I turn down the gains on that
channel, it doesn't seem to do much, or causes other volume issues.

Help!

In my research and hunting, I am starting to worry that the US-bought digium
cards have IMPEDENCE issues in the UK with the BT Lines etc?  That would
seem to explain why the echo is so incessant.  I have even tried changing
Echo Cancellers to MARK3.

Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian

Suggestions / Experiences in UK appreciated


--
Chris Earle
System Solutions Specialist,

- Original Message - 
From: "John Novack" <[EMAIL PROTECTED]>
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Wednesday, August 24, 2005 10:03 AM
Subject: Re: Connection TDM400P to UK PSTN


> The jacks on the TDM are ( incorrectly ) referred to as RJ45, correctly
> they are 8 position modular.
> The line, either in or out is on the two CENTER pins. NONE of the other
> 6 pins are used.
> Though I am not in the UK, from what I know you don't use the two center
> pins for a single line connection, so you will need to fashion some sort
> of adapter to connect. Frankly, using the two center pins ( A Bell
> System brain blizzard) wasn't the smartest idea. It makes the modular
> plug into, with the addition of just a little moisture, a really good
> spark gap when a ring signal or small induction of lightning is applied.
> I have seen many a modular plug turned black and useless  since the
> introduction of modular in the US in the early 70's
>
> Good luck
>
> John Novack
>
>
> Graham Kiff wrote:
>
> > I'm a complete Asterisk novice and have an installation based on the
> > [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>CD.
> >
> > I've installed my TDM400P with 2 x FXO & 2 x FXS, but every time I try
> > to dial out, I get a message "No circuits available".
> >
> > Can someone confirm the pinouts for connecting the FXO's to a UK BT
> > Line - I have RJ11 connectors on the back of my TDM400P card, so
> > ideally I'd like to know the pin mappings from a standard BT plug to
RJ11.
> >
> > Cheers
> > Graham
> >
> >
> >
> >___
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> >Asterisk-Users@lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Fritz card technology & German *

[Chris Earle \(CBL\)]

Thanks for all the posts everyone


So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I
would rather not have to apply patches just to get the two PCI cards to work
in the same box

The price difference between the cards you guys mentioned is interesting

I have also heard about BERONET isdn cards?  a single Beronet 4-channel card
would suffice I think?

Thing is, whatever the legacy system in place already is (this is not a
fresh operation) must have some sort of minor PBX in place, where all the
phones are plugged in.  So I would have to remove that and could use a TDM
card to plug the phones in?  These phones, isdn etc -- probably aren't
analog -- probably don't work with a TDM card right?
So I think what you were suggesting John is ISDN channel cards and a TDM in
the same machine?  with * just bridging calls between the two?

Interesting. :-S

Chris Earle

- Original Message - 
From: "John Daragon" <[EMAIL PROTECTED]>
To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Wednesday, January 18, 2006 6:45 AM
Subject: Re: [Asterisk-Users] Fritz card technology & German *


> Chris Earle (CBL) wrote:
> > Hi all,
> >
> > I've been working with * for a long time now, but only with analog
FXS/FXO
> > systems.
> > I am venturing towards setting up a box in Germany now and I believe
that
> > requires a Fritz card?  Do I even have to use the Fritz cards?  Why not
a
> > Digium card
>
> The AVM Fritz card is a single connection (2 x 64 kbps) passive ISDN
> card. It's well supported by chan_capi, but running more than one of
> them in a PC requires a driver patch.
>
> You can't use a Digium card because Digium doesn't make an ISDN2 card.
>
> >
> >   We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will
require 2
> > Fritz PCI cards (they have 1 port only).  Then there's some sort of
channel
> > bank that sends the calls out to the extensions.
> > Does this make any sort of sense?
>
> By 2 lines I guess you mean 4 channels ? i.e. 4 simultaneous calls ?  If
> you mean 2 channels, then you only need 1 fritz card.
>
>
> > Could someone confirm with me that this is the right direction to go -- 
ISDN
> > lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension
> > handsets..
>
> On the handset side you could use a couple of TDM4xx cards, or just use
> SIP phones.
>
> jd
>
> -- 
>
> John Daragon  [EMAIL PROTECTED]
> argv[0] limited   (Asterisk implementation & consultancy)
> Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
> v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127
>
>
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[Asterisk-Users] Fritz card technology & German *

[Chris Earle \(CBL\)]

Hi all,

I've been working with * for a long time now, but only with analog FXS/FXO
systems.
I am venturing towards setting up a box in Germany now and I believe that
requires a Fritz card?  Do I even have to use the Fritz cards?  Why not a
Digium card

  We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2
Fritz PCI cards (they have 1 port only).  Then there's some sort of channel
bank that sends the calls out to the extensions.
Does this make any sort of sense?

Could someone confirm with me that this is the right direction to go -- ISDN
lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension
handsets..


Thanks

--
Chris Earle
System Solutions Specialist,


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[Asterisk-Users] Re: Debian Vs Fedora

[Chris Earle]

Ya, going with whichever you know best is not a bad idea

Although I think more ppl have used Debian with success/little hassle than
with Fedora.
There are always little bugs with installing on the Fedora releases...

I ended up abandoning my fedora core 2 install in favour of sticking with
Debian.which I had on a couple of * servers already..keep is uniform
etc...

Good luck


"Syed Akbar" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Does anyone have any comments about using Debian stable release Vs Fedora
> for running Asterisk?
>
> Syed Akbar
>
> Alico Systems Inc
> www.alicosystems.com
> Tel: 562-436-1510
>
>
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[Asterisk-Users] Config files under CVS versioning system

[Chris Earle (CBL)]

Hi all,

I'm about to start building a bunch of asterisk servers with a team of
developers and I thought it would be a good idea to put each server's config
files under CVS so that we can keep track of changes, revert back etc...

Which files do you think I should include in the cvs modules?  just the
.conf files?

AND would it be possible to have the actual server copies be the versioned
copesi so all I'd have to do is a cvs update to install new changes?

Thoughs and suggestions appreciated,

Chris Earle
System Solutions Specialist


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Re: [Asterisk-Users] OT: list format vs newsgroup format

[Chris Earle (CBL)]

OE has the 'group messages by conversation' option, but I think I wrote it
off awhile ago because it didn't work so well or something.
maybe I'll give it a second chance

C  h  r  i  sE  a  r  l  e
System Solutions Specialist

- Original Message - 
From: "James Taylor" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 18, 2003 10:50 AM
Subject: RE: [Asterisk-Users] OT: list format vs newsgroup format


> So, other than Outlook for Win and Xfmail for Linux any recommendations
for those lost souls on the list?
>
> -- Original Message --
> From: "John Laur" <[EMAIL PROTECTED]>
> Reply-To: [EMAIL PROTECTED]
> Date: Fri, 18 Jul 2003 09:17:57 -0500
>
> >> Who else on here prefers the newsgroup/threaded approach?  If you
> >haven't
> >> already, check out news.gmane.org for mailing lists turned into
> >newsgroups
> >> readable by news readers...
> >
> >If you want threads, get a MUA that is capable of threading. Most are.
> >The In-Reply-To header makes mail threading on lists trivial (and you
> >can easily spot the people that hit reply and change the subject without
> >actually starting a new thread...) Maybe you just have not yet found
> >where to turn it on.
> >
> >> only problem being that this list requires list membership before
> >> posting
> >
> >Which is why the mailing list will probably continue to be used...
> >
> >John
> >
> >___
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> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> James Taylor
> [EMAIL PROTECTED]
> 903-793-1953
>
> --
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Re: [Asterisk-Users] OT: list format vs newsgroup format

[Chris Earle (CBL)]

- Original Message - 
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 18, 2003 10:16 AM
Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format


> On Fri, 2003-07-18 at 09:02, Chris Earle (CBL) wrote:
> > Agh
> >
> > I hate trying to sift through all these messages and keep track of the
> > various threads going on .
> >
> > Who else on here prefers the newsgroup/threaded approach?  If you
haven't
> > already, check out news.gmane.org for mailing lists turned into
newsgroups
> > readable by news readers...
>
> What you need is to get a decent mail reader. Those of us that complain
> regularly about people changing subjects in the middle of a thread
> already know the benefits of threaded email reading. Why bother with a
> newsgroup because you choose to stay on windows and use outbreak express
>

hmmI think I *do* realize the benefits of threaded email reading -- 
that's why I was supporting services like gmane! ;-)
I do realize that OE isn't really the best idea, but it works fairly well
with some proper setup (folders, rules etc).

> > only problem being that this list requires list membership before
> > posting
>
> And this is a good thing. Otherwise spammers only need the list address
> to spam us all, and you get this also on newsgroups. Right now the only
> risk is the fact that the email addresses we use here are archived
> publicly in an easy to harvest method. I think that is the most risk I
> wish to undertake.

Yes yes, I know it's a good thing, just didn't help my newsgroup
format/gmane solution...hehe
No complaints here about taking measures to prevent spam, it's more than
necessary IMO.




C  h  r  i  sE  a  r  l  e
System Solutions Specialist

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Re: [Asterisk-Users] OT: list format vs newsgroup format

[Chris Earle (CBL)]

what? you want to change the list into a message board? no no, newsgroups
are fine...

phpbb is nice though...

C  h  r  i  sE  a  r  l  e
System Solutions Specialist
- Original Message - 
From: "jltaylor " <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 18, 2003 10:35 AM
Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format


> "IF" there was a consideration for a change, I prefer:
>
> phpbb
>
> it's open source and easy to use.
>
> www.phpbb.com
>
> you can still get emails from the posts.
>
> ------ Original Message --
> From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
> Reply-To: [EMAIL PROTECTED]
> Date: Fri, 18 Jul 2003 10:02:22 -0400
>
> >Agh
> >
> >I hate trying to sift through all these messages and keep track of the
> >various threads going on .
> >
> >Who else on here prefers the newsgroup/threaded approach?  If you haven't
> >already, check out news.gmane.org for mailing lists turned into
newsgroups
> >readable by news readers...
> >
> >
> >only problem being that this list requires list membership before
> >posting
> >
> >
> >C  h  r  i  sE  a  r  l  e
> >System Solutions Specialist
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> James Taylor
> [EMAIL PROTECTED]
> 903-793-1953
>
> --
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
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[Asterisk-Users] OT: list format vs newsgroup format

[Chris Earle (CBL)]

Agh

I hate trying to sift through all these messages and keep track of the
various threads going on .

Who else on here prefers the newsgroup/threaded approach?  If you haven't
already, check out news.gmane.org for mailing lists turned into newsgroups
readable by news readers...


only problem being that this list requires list membership before
posting


C  h  r  i  sE  a  r  l  e
System Solutions Specialist

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Re: [Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes?

[Chris Earle (CBL)]

so, do you think that this upgrade is worth doing, or should I hold out a
bit longer (since I am not having any troubles right now) until the next
release?  I am assuming that there will probably still be a few more
upgrade releases...


C  h  r  i  sE  a  r  l  e
System Solutions Specialist

- Original Message - 
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, July 17, 2003 9:06 PM
Subject: [Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes?


>
> I see that there's now a 2.16.1 upgrade path for Cisco ATA-186
> devices, dated (variously) July 11 or July 14 2003.
>
> Here are some interesting bugs that claim to be fixed.  Most notable
> is CSCeb17953, at least from my perspective, as I've hit this bug
> before.
>
> CSCea42480 The Cisco ATA ignores the Require:100rel header and processes
call.
> CSCea69889 The Cisco ATA builds a 302 Moved Temporarily message
> incorrectly after receiving a NOTIFY message.
> CSCea93969 The Cisco ATA loses G.723 audio when call waiting occurs.
> CSCeb01064 The Cisco ATA From header domain value changes SRV record name.
> CSCeb17953 The Cisco ATA stops the registration process if it
> receives an unexpected response to a REGISTER request.
> CSCeb19228 The callback-on-busy feature does not work for calls to a PSTN.
> CSCeb23060 Upon receiving a 4xx response to a REGISTER request from a
> backup proxy, the Cisco ATA needs to continue retrying the request
> with the primary proxy .
> CSCeb24556 The Cisco ATA may fail to send a ring tone when acting as
> a transfer target in a blind transfer.
> CSCeb28218 The Cisco ATA, while in a call, detects audio from an incoming
call.
> CSCeb32210 When the SDP attribute a=fmtp appears before the attribute
> a=rtpmap , the Cisco ATA will not send out-of-band DTMF digits.
> CSCeb35955 Attended call transfers occur even when this feature is
> disabled via the PaidFeature configuration parameter.
> CSCeb36752 Call forwarding does not work when the Cisco ATA detects a
> busy signal.
> CSCeb37037 The Cisco ATA stops registering after a 2.16 upgrade is
performed.
> CSCeb37043 The call-waiting default user setting cannot be controlled
> by the CallFeatures configuration parameter when the Cisco ATA
> obtains its configuration file from the TFTP server.
> CSCeb40099 The Cisco ATA plays an incorrect tone after unconditional
> call forwarding is enabled or disabled.
> CSCeb44406 Change the behavior of the Cisco ATA to not remove all
> registrations.
>
> Full information can be found here:
>
>
http://www.cisco.com/en/US/products/hw/gatecont/ps514/prod_release_note09186a00801a2519.html
>
>
> JT
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[Asterisk-Users] Asterisk on Cygwin?

[Chris Earle \(CBL\)]

Hey all,

quick question: does asterisk work okay in a Cygwin environment?

I want to install it on my cygwin setup for local testing/demoing and save
me the hassle of using a pure linux machine

As long as it doesn't take a huge huge performance hit from running out of
Cygwin, then I'll have a go there for a start

confirmation appreciated!
thanks



-- 
C  h  r  i  sE  a  r  l  e
System Solutions Specialist


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