[asterisk-users] CLI not showing DTMF

2007-10-18 Thread Volker Sauer
Hi,

after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore,
even at high debug level. Do I need to activate that?

Regards
Volker
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Re: [asterisk-users] Refrigerator Alarms

2007-10-18 Thread Per Jessen
Balu Raman wrote:

 Omar,
 I am hoping that there may be some temp sensor interface that can be
 routed to a pc and if the temp falls out of a range, I can have this
 event call someone. I know what to do in asterisk to make a call. I
 have to do some research. may be,  someone has already done a similar
 thing. Has to be event driven.

Here's what we do - even it's not asterisk-related - temperatures are
monitored/polled using Maxim/Dallas DS1820s devices.  These are cheap
and the size of a transistor.
When/if certain thresholds are exceeded, an email is sent to our central
mail-server where it is turned into an SMS.  The same email could just
as easily be turned into a call file and dropped into to the
appropriate asterisk directory. 
We expect to start using SNMP traps instead of the email, but the
principle is pretty much the same. 



/Per Jessen, Zürich

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Re: [asterisk-users] Issues with Zaptel 1.4.5.1

2007-10-18 Thread Alan Lord
Tzafrir Cohen wrote:
snip /... is required to properly start zaptel. It will also run 
ztcfg. Otherwise
 users run into issues where misconfigured zaptel.conf fails loading of a
 module. That is a buggy behaviour.
 
 If your card is an analog one, take a look at http://bugs.digium.com/7613
 and tell me what you think.
 
 Something similar for digital spans would require more information in
 sysfs.
 

So do I understand this correctly? I can use this patch to alleviate the 
need for a userspace init.d start-up script?

I just patch the zaptel module source and rebuild?

Does it read /etc/zaptel.conf and thus load the zone data?

I only have an x100p so this could be a useful patch for me.

Alan

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[asterisk-users] Relaying calls to another SIP extension

2007-10-18 Thread Paul Campbell
Hi,

 

I am learning Asterisk for a small project.  

 

At this stage I have an AsteriskNOW system running locally. I can call
SIP phone to SIP phone fine, the operator and voice mail work fine,
except some stuttering (probably caused by it running in MS-VPC)

 

What I need to figure out is...

 

I have an automated response telephony server (Voice Media Server)
available via a SIP URI like:

 

[EMAIL PROTECTED]

 

If I place this into the SIP phone, X-Lite the call goes through fine...
but not through my Asterisk.

 

What I need to do is to add this service 'through' my Asterisk.  So when
a user calls a certain extension through Asterisk it 'rings' the
extension [EMAIL PROTECTED] as if it were a standard SIP phone.

 

That is stage 1.

 

Stage 2 is for the media server handling [EMAIL PROTECTED] to be able to
forward the call onto another SIP phone, allowing it to drop the call
completely and the call tromboning or bridging to happen in the Asterisk
PBX and not take up 'lines' on the media server.  (Eventually both SIP
ends may become PSTN's).

 

I'm having trouble deducing how to do stage 1.  In asteriskNow I don't
see a way to add a calling rule for this.  I tried adding a service
provider foo.bar.com and a calling rule to send all calls for
extension 6002 to that provider, but all I get is Service unavailable.

 

With the Asterisk-docs site down I'm finding it tough going.

 

Thanks for any pointers you can give me.

Paul Campbell  

 




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Re: [asterisk-users] asterisk incomming call huntgrroup

2007-10-18 Thread Lenz

Try this: http://astrecipes.net/index.php?n=42
l.



On Thu, 18 Oct 2007 06:43:11 +0200, satish patel  
[EMAIL PROTECTED] wrote:

 I am new in asterisk world can u shortly explian how to create queue and  
 how to work this ?

 David Gomillion [EMAIL PROTECTED] wrote:
 On 10/17/07, satish patel [EMAIL PROTECTED] wrote: Dear  
 all
   I want to configure Huntgroup for my company like i call  
 on 1100 extention i will transfer to avalible group extention i got some  
 document on voip-info website but this is not working for me
  http://www.voip-info.org/wiki/view/Asterisk+Hunting+Groups+for+incoming+calls

 Why not just use a queue?


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Re: [asterisk-users] Cisco phones with Asterisk

2007-10-18 Thread Olivier
This have been discussed a couple of weeks ago in this list.
You should find useful and detailed answers in archives.

Regards
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[asterisk-users] Is anyone successfully using IMAP storage

2007-10-18 Thread Olivier
Hi,

From personal experience, Asterisk 1.4 IMAP storage seems broken and
unusable.
Is anyone using it successfully ?

This kind of poll would be very useful to estimate is a code rewrite has a
chance to disturb a running system.
I we get no successful report, it would help developers to consider a code
rewrite as patch instead of a new feature.

So please, do not hesitate to report successful or unsuccessful or never
tried use of IMAP storage, it will increase our chances to get this feature
in 1.4 branch and not wait for 1.6.

Cheers
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Re: [asterisk-users] Crash related to asterisk -rx ?

2007-10-18 Thread Atis Lezdins
On Thursday 18 October 2007 04:47:14 Jean-Denis Girard wrote:
 Hi list,

 Last Friday, an Asterisk server became unresponsive after ~8,5 months of
 smooth operation (~32 calls). Server did reply to pings, but no ssh,
 no more console login. Also Asterisk no longer took calls, but ISDNguard
 watchdog was still alive. Looking at the logs after reboot, I could not
 find anything significant, except in a file created by the following
 command via a cron job:

 date  /var/log/asterisk/calls.log ; asterisk -rx show channels
 concise  /var/log/asterisk/calls.log

 Two days before the crash, the calls.log file started to be filled with
 the Asterisk console messages. I suspect this is what caused the server
 crash. Anybody seen this before, is this a known problem with asterisk
 -rx commands?

Yup, it's also a problem for me, but it haven't ever crashed server. It just 
makes specific remote process unresponsive. There's a patch for 1.4, but i 
guess it wouldn't be hard to backport it for 1.2

http://bugs.digium.com/view.php?id=10847

you might also want the one mentioned in comments:

http://bugs.digium.com/view.php?id=10888

Regards,
Atis

-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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[asterisk-users] Softphone that emulates Skype API ?

2007-10-18 Thread Cosmin Prund
There's a large number of gadgets one can buy that work with Skype
through the API. One of the things I'm interested right now is the
ability to properly use a mobile phone headset with a SIP/IAX softphone.

 

Is there an softphone that emulates the Skype API?

Are there legal implications in writing an softphone that emulates the
Skype API?

Should I just give up and buy a Siemens DECT phone that supports a
bluetooth headset?

 

--

Thanks,

Cosmin Prund 

 

 

 

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Re: [asterisk-users] phone as control interface(was99bottlesofbeer)

2007-10-18 Thread marcotasto
Hi All, 
sorry if I post again this e-mail but I think the first one was lost.

I don't know if this is OT but I'm working in my spare time at  a small 
hardware project that match to what's requested below.
It's a board with Input/Output capabilities and 10Mbps ethernet interface. It 
has Microchip software TCP/IP stack on it. 
Being at a very beginning stage, you can see a little preview (and hopefully 
play with an online prototype) at this address: 

http://www.auto-matica.com/index.php?id=16

I've used it in the past to open a garage door when calling a private extension 
with Asterisk PBX and a little perl-AGI glue.

I hope could interest to someone and I'm open to any suggestion/collaborations.

Thank you and bye,
Marco.


- Original Message -
From: John Faubion [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 9:46 AM
Subject: Re: [asterisk-users] phone as control interface(was99bottlesof
beer)


 has anyone actually been satisfied with the performance of these
 powerline signalling devices ?

 Universal Powerline Bus is a vast improvement over the original X10. I
 believe the X10 devices used a 4V signal during the zero crossing point of
 the AC voltage to transmit 1 bit. Needless to say this made X10 slow,
 susceptible to line noise and not very reliable. IIRC X10 is only 75%
 reliable. By contrast, UPB is 20-40 times faster, it uses a higher
 signaling
 voltage so line noise isn't a big factor any more and has the advantage of
 controlling 250 times more devices than X10. This will help to prevent
 stray
 signals from the neighbors controller from accidentally controlling your
 devices. UPB is supposed to be 99.9% reliable with a latency of less than
 100 milliseconds. Granted that still leaves a tenth of a percent of
 uncertainly. However that is without resorting to filters, couplers and
 the
 like.

 Granted in an existing situation there may not be a way to run more
 wires, but I evaluated them a while back and decided to stay away.

 You may want to take another look at them. Just like Asterisk has made
 great
 strides since the release of 0.7, UPB has brought the quality level way
 up.
 Of course this higher quality also has a higher price.

 John


Anyone on the list interested on working on a project where we can create
devices that work over Ethernet ?




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[asterisk-users] A linksys SPA921 behind NAT and firewall

2007-10-18 Thread Per Jessen
I've got someone sat in a home-office with an SPA921 behind NAT, and
most probably a firewall.  I've got a STUN-server running, and calling
in from the SPA921 to our Asterisk box works fine - though I had to
open the firewall for UDP traffic on port 1-2. 

Calling from our Asterisk to the SPA921 doesn't work.  I'm guessing this
is due to the NAT/firewall on the other side, coz' how would it know
that UDP-traffic to SPA publicIP:5060 needs to be delivered to
192.168.x.x:5060 ?  



/Per Jessen, Zürich

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Re: [asterisk-users] phone as control interface(was99bottlesofbeer)

2007-10-18 Thread Lenz

Hello Marco,
could you explain how you did the interfacing to the Asterisk PBX? does  
your prototype speak SIP to receive commands?
Thanks
l.




On Thu, 18 Oct 2007 12:27:33 +0200, marcotasto [EMAIL PROTECTED]  
wrote:

 Hi All,
 sorry if I post again this e-mail but I think the first one was lost.

 I don't know if this is OT but I'm working in my spare time at  a small  
 hardware project that match to what's requested below.
 It's a board with Input/Output capabilities and 10Mbps ethernet  
 interface. It has Microchip software TCP/IP stack on it.
 Being at a very beginning stage, you can see a little preview (and  
 hopefully play with an online prototype) at this address:

 http://www.auto-matica.com/index.php?id=16

 I've used it in the past to open a garage door when calling a private  
 extension with Asterisk PBX and a little perl-AGI glue.

 I hope could interest to someone and I'm open to any  
 suggestion/collaborations.

 Thank you and bye,
 Marco.


 - Original Message -
 From: John Faubion [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 16, 2007 9:46 AM
 Subject: Re: [asterisk-users] phone as control interface(was99bottlesof
 beer)


 has anyone actually been satisfied with the performance of these
 powerline signalling devices ?

 Universal Powerline Bus is a vast improvement over the original X10. I
 believe the X10 devices used a 4V signal during the zero crossing point  
 of
 the AC voltage to transmit 1 bit. Needless to say this made X10 slow,
 susceptible to line noise and not very reliable. IIRC X10 is only 75%
 reliable. By contrast, UPB is 20-40 times faster, it uses a higher
 signaling
 voltage so line noise isn't a big factor any more and has the advantage  
 of
 controlling 250 times more devices than X10. This will help to prevent
 stray
 signals from the neighbors controller from accidentally controlling your
 devices. UPB is supposed to be 99.9% reliable with a latency of less  
 than
 100 milliseconds. Granted that still leaves a tenth of a percent of
 uncertainly. However that is without resorting to filters, couplers and
 the
 like.

 Granted in an existing situation there may not be a way to run more
 wires, but I evaluated them a while back and decided to stay away.

 You may want to take another look at them. Just like Asterisk has made
 great
 strides since the release of 0.7, UPB has brought the quality level way
 up.
 Of course this higher quality also has a higher price.

 John


 Anyone on the list interested on working on a project where we can create
 devices that work over Ethernet ?




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[asterisk-users] Blocking collect calls in Brazil

2007-10-18 Thread Carlos Barros
Hi list.

I was googling about this subject and found a message to this list
asking the same thing I want, but see no response. So, I'm reposting
it in case someone has a solution for this.
I'd like to know if there's some way to detect (not block) a collect
call in a ISDN E1? I need this cause I have my own billing system and
I need to charge my customers for collect calls they receive.
Currently I have to get the PSTN billing and locate all collect calls
and then enter it manually. I want to detect these calls in asterisk
so I can automatically charge my customers.

For Brazilians in the list, I have a Embratel E1 - ISDN.

Thanks

Carlos Barros

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Re: [asterisk-users] Asterisk on USB Flash?

2007-10-18 Thread Michael Graves
On Thu, 18 Oct 2007 00:32:44 -0400, Brian Capouch wrote:

shadowym wrote:
 Size/Speed/write cycles have gone way up, price has gone way down.  More
 common than CompactFlash and no need for an adapter.  So is it feasible to
 run an Asterisk server on something like this?  With a MTBF of 1million
 write cycles coupled with dynamic wear management on a 4Gig USB drive,
 lifetime is a non-issue.  Just wondering how well it works, if it works.  
 

My main server for my home and teensy business runs on a Netgear 
WGT634U running SVN-trunk under openWRT.

The Asterisk binary sits on the machine's flash, but all the modules, 
prompt files, voicemail, etc., goes to the flash.

It works just fine.  This system doesn't get a huge amount of voicemail, 
so I don't know about how long it would be before the wear issue would 
surface (pun?).

I know it works just fine.  I've been doing this now for almost two 
years, although with various versions of embedded Linux and various 
versions of Asterisk.


Yep, me too. I've been running Asterisk on Soekris Net4801 and H-P
T5700 booting from either CF or USB key for just over two years. No
problems at all for a small office. In fact, its kinda a waste to have
to buy 1 GB+ USB keys when it really needs  64 MB. 

In fact, my Net4801 box is using a 64 MB CF card that I receyled from
my first digital camera. That card must be 7 years old. Still works.
Not much VM so not much wear.

Michael
--
Michael Graves
mgravesatmstvp.com
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245



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[asterisk-users] sorta OT: Bounty for Click to Call plugin for IE

2007-10-18 Thread Michael Graves
I'm in process of transitioning a number of offices to a hosted virtual
pbx from Junction Networks. It's a combination of OpenSER and Asterisk.
They have a nice click-to-call extension for Firefox, but I need the
equivalent for IE so that it can work with our CRM system. Junction
told me that they have a bounty on offer for this if someone's
interested in doing the work.

Would the availability of the Firefox code make it easier to do an
ActiveX implementation?

Any takers?

Michael

--
Michael Graves
mgravesatmstvp.com
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245



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[asterisk-users] BBC on Atserix

2007-10-18 Thread Cartwright, Dave
Just for fun.

 

http://news.bbc.co.uk/1/hi/magazine/7049642.stm

 

Dave


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Re: [asterisk-users] Refrigerator Alarms

2007-10-18 Thread Dovid B
Kevin,
What kind of device are you using on the fridge ?

Dovid
  - Original Message - 
  From: Kevin Withnall 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Thursday, October 18, 2007 12:04 AM
  Subject: Re: [asterisk-users] Refrigerator Alarms


  We use similar things here for issues like our generator battery voltage 
monitoring. We just have a relay going into our alarm system and as asterisk 
monitors our alarms it initiates emails or calls out. The alarm system is also 
linked into a seperate SMS unit for emergency backup so we also get SMS when 
any alarm goes off.

  My basic alarmreceiver scripts are available at 
http://kevin.withnall.com/2007/07/09/asterisk-alarm-receiver-using-triggers-mysql5/
 if anyone wants them.

  --
  Kevin Withnall http://kevin.withnall.com/
  ILB Computing http://www.ilb.com.au
  PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081
  Please consider the environment before printing this e-mail







From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, 18 October 2007 7:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Refrigerator Alarms


The refrigerators will have external outputs to trip relays (even if your 
customer doesn't know that), ask for the number of their refrigeration mechanic 
he will tell you how to get electrical/relay outputs for the alarms.

 

These are then connected to asterisk via an interface board so when 
something trips it results in an event in asterisk (check out the asterisk at 
home X10 configurations for some ideas), this will then result in a call with a 
number (or even better a recording being played), eg meat fridge number 1  
at  6  degrees celcius

 

That's pretty much your answer J

 

Regards,

Dean Collins
[EMAIL PROTECTED] 
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).




From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balu Raman
Sent: Wednesday, 17 October 2007 4:47 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Refrigerator Alarms

 

Hi,
I want asterisk to call a person on the phone for monitoring the 
refrigerator storing vaccines.
I am clueless where to look. Can someone clue me in ?
Thanks,
balu raman



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Re: [asterisk-users] Refrigerator Alarms

2007-10-18 Thread Drew Gibson

Hi Balu,

http://www.digitemp.com/ has all the info you need. Cost of hardware 
around US$70. Will give you pretty fridge temperature graphs too!
You can easily hack some of Brian's scripts to do different levels of 
temp alarm and trigger calls in Asterisk.


regards,

Drew


Balu Raman wrote:

Omar,
I am hoping that there may be some temp sensor interface that can be
routed to a pc and if the temp falls out of a range, I can have this
event call someone. I know what to do in asterisk to make a call. I
have to do some research. may be,  someone has already done a similar
thing. Has to be event driven.
Thanks,
balu raman

On 10/17/07, Omar A. Sabek [EMAIL PROTECTED] wrote:
  

Balu,

Do you want events passed to Asterisk from the refrigerator? Or does a
reminder type phone call need to be placed on an interval? Please be
more specific, since this sounds like a special purpose refrigerator,
does it have any way of passing events to an external device?

Omar A. Sabek

On 10/17/07, Balu Raman [EMAIL PROTECTED] wrote:


Hi,
I want asterisk to call a person on the phone for monitoring the
refrigerator storing vaccines.
I am clueless where to look. Can someone clue me in ?
Thanks,
balu raman

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Re: [asterisk-users] phone as control interface(was99bottlesofbeer)

2007-10-18 Thread marcotasto
Hi Lenz.
What I did to interface asterisk with the door opener was to implement, in the 
board, a custom embedded server that receives and parses a set of UDP packets 
containing a known data in it. 
In the dialplan I then call a perl AGI script that sends UDP packets in the 
correct sequence and with proper contents inside.
I've implemented in this way to be able to open the door not only by calling an 
internal extension but even through a dedicated external pushbutton connected 
to a second board that sends the same UDP packet sequence when the pushbutton 
is pressed.

I did some other experiments trying to embed a very lite SIP layer (written 
from scratch to be the more embeddable as possible) and I was able to register 
to my Asterisk PBX and to answer to OPTION packets (sent by the PBX) to be 
qualified as a valid SIP channels... but it's today only an experiment because 
I never had time to terminate it. :-(
SIP is complex and to write a SIP compliant layer is a very time consuming 
stuff.

Another think that could be done is to use the already working HTTP server 
layer. Think about sending an HTTP GET or POST, again, with an external perl 
AGI script.

Thank you and bye.

Marco Signorini.


  Original Message 
 Subject: Re: [asterisk-users] phone as control interface(was99bottlesofbeer)
 From:Lenz [EMAIL PROTECTED]
 Date:Thu, October 18, 2007 1:15 pm
 To:  Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 --
 
 
 Hello Marco,
 could you explain how you did the interfacing to the Asterisk PBX? does
 your prototype speak SIP to receive commands?
 Thanks
 l.
 
 
 
 
 On Thu, 18 Oct 2007 12:27:33 +0200, marcotasto [EMAIL PROTECTED]
 wrote:
 
  Hi All,
  sorry if I post again this e-mail but I think the first one was lost.
 
  I don't know if this is OT but I'm working in my spare time at  a small
  hardware project that match to what's requested below.
  It's a board with Input/Output capabilities and 10Mbps ethernet
  interface. It has Microchip software TCP/IP stack on it.
  Being at a very beginning stage, you can see a little preview (and
  hopefully play with an online prototype) at this address:
 
  http://www.auto-matica.com/index.php?id=16
 
  I've used it in the past to open a garage door when calling a private
  extension with Asterisk PBX and a little perl-AGI glue.
 
  I hope could interest to someone and I'm open to any
  suggestion/collaborations.
 
  Thank you and bye,
  Marco.
 
 
  - Original Message -
  From: John Faubion [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Tuesday, October 16, 2007 9:46 AM
  Subject: Re: [asterisk-users] phone as control interface(was99bottlesof
  beer)
 
 
  has anyone actually been satisfied with the performance of these
  powerline signalling devices ?
 
  Universal Powerline Bus is a vast improvement over the original X10. I
  believe the X10 devices used a 4V signal during the zero crossing point
  of
  the AC voltage to transmit 1 bit. Needless to say this made X10 slow,
  susceptible to line noise and not very reliable. IIRC X10 is only 75%
  reliable. By contrast, UPB is 20-40 times faster, it uses a higher
  signaling
  voltage so line noise isn't a big factor any more and has the advantage
  of
  controlling 250 times more devices than X10. This will help to prevent
  stray
  signals from the neighbors controller from accidentally controlling your
  devices. UPB is supposed to be 99.9% reliable with a latency of less
  than
  100 milliseconds. Granted that still leaves a tenth of a percent of
  uncertainly. However that is without resorting to filters, couplers and
  the
  like.
 
  Granted in an existing situation there may not be a way to run more
  wires, but I evaluated them a while back and decided to stay away.
 
  You may want to take another look at them. Just like Asterisk has made
  great
  strides since the release of 0.7, UPB has brought the quality level way
  up.
  Of course this higher quality also has a higher price.
 
  John
 
 
  Anyone on the list interested on working on a project where we can create
  devices that work over Ethernet ?
 
 
 
 
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 http://queuemetrics.com
 

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Re: [asterisk-users] CLI not showing DTMF

2007-10-18 Thread Jared Smith
On Thu, 2007-10-18 at 09:25 +0200, Volker Sauer wrote:
 after I upgraded from 1.2 to 1.4.13 the CLI does not show DTMF anymore,
 even at high debug level. Do I need to activate that?

You need to enable DTMF debugging in logger.conf, then type logger
reload at the Asterisk CLI for those changes to take effect.


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[asterisk-users] Automating blacklists

2007-10-18 Thread Brian Hutchinson
Hi,

I've been reading all I can on Google (and Asterisk TFOT book) looking for
ideas on how to implement an automated blacklist feature.

I would like to automatically blacklist a incoming number based on timestamp
and count information.

For example, if I get a prank call from the same number 5 times within 15
minutes, I want my dialplan to automatically blacklist this number.

Should I be looking at AGI to do something like this?

Thanks for any ideas or pointers!
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[asterisk-users] Mandriva 2008

2007-10-18 Thread Doug Lytle
Just a note for those of you trying to install Asterisk 1.4.x under 
Mandriva 2008.

I wasn't able to get make menuselect to work.  It kept telling me that I 
didn't have ncurses installed (I have  ncurses-5.6-1) even though the 
configure script said it found it.  I eventually went into the 
menuselect directory and did a ./configure and make.  After I did that, 
I went back into the Asterisk root and did a make menuselect and 
everything worked.

Doug

-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.



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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Jared Smith
On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote:
 I would like to automatically blacklist a incoming number based on
 timestamp and count information. 
 
 For example, if I get a prank call from the same number 5 times within
 15 minutes, I want my dialplan to automatically blacklist this number.
 
 Should I be looking at AGI to do something like this?

You could do it with an AGI, or with dialplan logic and the AstDB
database.  If you use the AstDB database to store blacklisted numbers,
you can also use the BLACKLIST dialplan function to check to see if a
given number is blacklisted or not.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Refrigerator Alarms

2007-10-18 Thread Jim Canfield
Balu Raman wrote:
 Hi,
 I want asterisk to call a person on the phone for monitoring the 
 refrigerator storing vaccines.
 I am clueless where to look. Can someone clue me in ?
OWFS for sure!  Here is a screenshot of a program I created a couple 
years ago to monitor refrigerators/warmers. It would be trivial to have 
the alarm script call a phone.  I have been out of the OWFS loop for a 
while, but I'm pretty sure there is native alarming and set points now.

http://sourceforge.net/project/screenshots.php?group_id=85502ssid=33253

Here the owfs website:

http://www.owfs.org/

Msg me off list if you have any questions.

-jc





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[asterisk-users] Limit number of times a call can be forwarded

2007-10-18 Thread Don Pobanz
We have had a few different times when a user has forwarded their phone
to himself. This has overloaded the communications to our operator panel
(FOP). One user should not be able to effect the whole phone system! 

Is there a way that the number of times that a call can be forwarded
could be limited like to 10 or even 100? Then even if a user does
something stupid like forwarding their calls to himself, it wouldn't
cause problems for others. 

Don Pobanz

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[asterisk-users] IAX2: Calls answered before extension is tested?

2007-10-18 Thread Alan Lord
[Sorry if this arrives more than once. I have sent this twice and it 
never arrived, despite other messages getting to the list O.K.]
---

Hello,

I would like an incoming caller to be able to choose from the menu 
options in my extension.conf below. Once They have dialled the 
appropriate digit, * should call two extensions simultaneously: one SIP 
phone on this * server, and one over a working IAX2 link. If either 
party answers, great. If neither party answers or both are 
busy/unavailable go to relevant voicemail box.

This *almost* works... ;-)

The issue is the call that goes down the IAX2 channel always seems to 
get answered, then cleared down almost straight away. It looks like the 
remote * server is accepting the incoming call sending a connected 
message back, thereby completing the Macro, and only *then* checking if 
the extension is actually available.

Here's the last bit of the log (I've edited the IP address) - we are 
both deliberately NOT answering our phones...

  Executing [EMAIL PROTECTED]:1] Macro(SIP/101-081d1050, 
belllord|SIP/101IAX2/alanb/201|tolc) in new stack
 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-081d1050, 
SIP/101IAX2/alanb/201|10|tr) in new stack
 -- Called 101
 -- Called alanb/201
[Oct 17 16:09:47] WARNING[2836]: channel.c:2634 ast_indicate_data: 
Unable to handle indication 3 for 'SIP/101-081d1050'
 -- SIP/101-081d4fc0 is ringing
 -- Call accepted by 80.XXX.XX.XX (format alaw)
 -- Format for call is alaw
 -- IAX2/alanb-3 answered SIP/101-081d1050
[Oct 17 16:09:47] NOTICE[2836]: cdr.c:434 ast_cdr_free: CDR on channel 
'SIP/101-081d4fc0' not posted
[Oct 17 16:09:47] DEBUG[1419]: chan_iax2.c:7435 socket_process: 
Immediately destroying 3, having received hangup
[Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3176 iax2_hangup: We're 
hanging up IAX2/alanb-3 now...
[Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3191 iax2_hangup: Really 
destroying IAX2/alanb-3 now...
 -- Hungup 'IAX2/alanb-3'
   == Spawn extension (macro-belllord, s, 1) exited non-zero on 
'SIP/101-081d1050' in macro 'belllord'
   == Spawn extension (macro-belllord, s, 1) exited non-zero on 
'SIP/101-081d1050'

And here's the relevant bits of my extension.conf

[globals]
ALANL=SIP/101 ; My Soft Phone
ALANB=IAX2/alanb/201 ; Alan's Extension

[main_menu] ; Test Dialplan for IVR
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5) ; Max time between digits
exten = s,n,Set(TIMEOUT(response)=15) ; Max time to wait
exten = s,n,Wait(1)
exten = s,n,Background(welcome-to-bell-lord)
exten = s,n(resume),Background(press-3-for-tolc) ; Short dialogues,
exten = s,n,Background(press-4-for-fondoo) ; rather than one long one
exten = s,n,Background(press-5-for-arrowtees) ; might need to change
exten = s,n,Background(press-6-for-gen-enq) ; frequently.
exten = s,n,WaitExten()

exten = 3,1,Goto(tolc,s,1) ; Dial 3 For The Open Learning Centre
exten = 4,1,Goto(fondoo,s,1) ; Dial 4 for Fondoo Internet
exten = 5,1,Goto(arrowtees,s,1) ; Dial 5 for ArrowTees
exten = 6,1,Goto(gen_enq,s,1) ; For all other enquiries press 6

exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(resume)

exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup() ; Might change this section to go to [gen_enq] 
voicemail rather than just hangup.

[tolc]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) ; Calls the 
belllord Macro with the channel(s) to dial and the current context (for 
business voicemail)

[fondoo]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})

[arrowtees]
exten = s,1,Macro(belllord,${ALANL},${CONTEXT})

[gen_enq]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})

; Call with Macro(belllord,channel,vmbox)
[macro-belllord] ; Uses macro and DIALSTATUS for local devices
exten = s,1,Dial(${ARG1},10,tr)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED],u) ; business is the 
voicemail context, ${ARG2} is the context from which this call came
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED],b)
exten = _s-.,1,Goto(s-NOANSWER,1)

==

Can anyone see where the problem is? Or suggest a better way?

Many thanks.

Alan


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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Brian Hutchinson
Is there a function to write the timestamp of the first call?  I started
thinking about AGI and PHP/MySQL since that is what I'm familiar with.  I
couldn't find methods to write timestamp info to AstDB or if I could ... how
to read it back and compare it to time now to decide to increment my
counter and have the dial plan decide to allow the call or not.

On 10/18/07, Jared Smith [EMAIL PROTECTED] wrote:

 On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote:
  I would like to automatically blacklist a incoming number based on
  timestamp and count information.
 
  For example, if I get a prank call from the same number 5 times within
  15 minutes, I want my dialplan to automatically blacklist this number.
 
  Should I be looking at AGI to do something like this?

 You could do it with an AGI, or with dialplan logic and the AstDB
 database.  If you use the AstDB database to store blacklisted numbers,
 you can also use the BLACKLIST dialplan function to check to see if a
 given number is blacklisted or not.


 --
 Jared Smith
 Community Relations Manager
 Digium, Inc.


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[asterisk-users] receiving fax over sip extension.

2007-10-18 Thread Sanspareils Greenlans
Sir,

I am having runing asterisk 1.4 server which is runing without any problem now 
i want to receive fax over sip extension. how it is possible and what the 
change i have make in extensions.conf.

Thanks

Rajeev.

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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Jared Smith
On Thu, 2007-10-18 at 16:51 +0300, Brian Hutchinson wrote:
 Is there a function to write the timestamp of the first call?

You can use the built-in channel variable ${EPOCH}, which will give you
the current time in Unix timestamp format (number of seconds since Jan
1, 1970).  Then you can do something like:

exten = 123,n,Set(DB(myblacklist/${CALLERID(num)}/lastcall)=${EPOCH}))

This obviously doesn't solve the entire problem for you, but should at
least help you get moving in the right direction.

-- 
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Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] BBC on Atserix

2007-10-18 Thread Tzafrir Cohen
On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
 Just for fun.
 
 http://news.bbc.co.uk/1/hi/magazine/7049642.stm

It's Asterix != Asterisk. Though named after *.

-- 
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icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Mandriva 2008

2007-10-18 Thread Jared Smith
On Thu, 2007-10-18 at 09:08 -0400, Doug Lytle wrote:
 I wasn't able to get make menuselect to work.  It kept telling me that I 
 didn't have ncurses installed (I have  ncurses-5.6-1) even though the 
 configure script said it found it.  I eventually went into the 
 menuselect directory and did a ./configure and make.  After I did that, 
 I went back into the Asterisk root and did a make menuselect and 
 everything worked.

Yes, that's not an uncommon problem, and it's no no way limited to just
Mandriva.  What happens is that the menuselect autoconf doesn't get
re-run, so it never learns that you've installed the ncurses libraries.

I always suggest people do a make distclean after installing any
additional libraries, and then go on to the ./configure step.  This
will ensure that the menuselect autoconf stuff gets cleaned out.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Lenz

It's not technically complex to do - you can probably use the astdb for  
that, or store all incoming numbers with timestamp in MySQL and run  
something like:

SELECT count(*)  5 AS blacklisted
 FROM incoming_calls
WHERE callerid = 12345
AND timestamp  DATE_SUB( NOW(), INTERVAL 15 MINUTE )

you should be very well aware of the risks that can stem from such a  
program - in case of bugs, or anomalous situations, you might end up  
blacklisting somebody who actually needs to call in.

I hope this helps
l.










On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson  
[EMAIL PROTECTED] wrote:

 Hi,

 I've been reading all I can on Google (and Asterisk TFOT book) looking  
 for
 ideas on how to implement an automated blacklist feature.

 I would like to automatically blacklist a incoming number based on  
 timestamp
 and count information.

 For example, if I get a prank call from the same number 5 times within 15
 minutes, I want my dialplan to automatically blacklist this number.

 Should I be looking at AGI to do something like this?

 Thanks for any ideas or pointers!



-- 
Loway Research - Home of QueueMetrics
http://queuemetrics.com

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[asterisk-users] Issues with making calls

2007-10-18 Thread Pablo Almido
Hi List,

I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server


[Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
channel type registered for 'Zap'
[Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
Unable to create channel of type 'Zap' (cause 66 - Channel not
implemented)
  == Everyone is busy/congested at this time (1:0/0/1)


My sources are:

libpri-1.4.1.tar.gz
zaptel-1.4.5.1.tar.gz
asterisk-1.4.11.tar.gz
asterisk-addons-1.4.2.tar.gz
asterisk-perl-0.10.tar.gz



I have 1/2 E1  from my provider telephony,  my configuration is

[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = us
defaultzone=us





#cat /etc/asterisk/zapata.conf
[channels]
context=default
switchtype = euroisdn
pridialplan = unknown
signalling = pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=no
amaflags=documentation
musiconhold=default


;Configure Channels
group=0
callgroup=0
pickupgroup=0
channel = 1-4
group=1
callgroup=1
pickupgroup=1
channel = 5-8
group=2
callgroup=2
pickupgroup=2
channel = 9-12
group=3
callgroup=3
pickupgroup=3
channel = 13-14
group=4
callgroup=4
pickupgroup=4
channel = 15


I have could make calls but, after of some minutes my server is hung,
suggestions are welcome. Thanks for any help in advance.

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Re: [asterisk-users] Is anyone successfully using IMAP storage

2007-10-18 Thread Rob Hillis
Not using IMAP storage here, although that was one of the primary 
drivers for upgrading to Asterisk 1.4.  Why?


The short answer is that it's too confining.  There are too many caveats 
that don't fit into our existing IMAP structure and make the entire 
project rather iffy.  Security is a concern too - we don't want to have 
one set of credentials that have absolute access to everyone's email 
from the CEO down.  I suspect we /could/ make it work, but the thought 
of experimenting on our live IMAP server sends chills down my spine.  
(we don't have the budget or time to duplicate our mail server to allow 
fiddling on a non-production system)


The idea is brilliant.  The weird dependency on having a 
compiled-but-not-installed UW-IMAP isn't - I would have thought that 
using a library such as libvmime would have made more sense.


Olivier wrote:

Hi,

From personal experience, Asterisk 1.4 IMAP storage seems broken and 
unusable.

Is anyone using it successfully ?

This kind of poll would be very useful to estimate is a code rewrite 
has a chance to disturb a running system.
I we get no successful report, it would help developers to consider 
a code rewrite as patch instead of a new feature.


So please, do not hesitate to report successful or unsuccessful or 
never tried use of IMAP storage, it will increase our chances to get 
this feature in 1.4 branch and not wait for 1.6.
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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Brian West
Make sure chan_zap.so is loaded.

/b

On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote:

 Hi List,

 I am from Peru, I have installed an asterisk server in my company with
 digium card E1 TE120P, I am having issues when i make calls, here the
 error from my server


 [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
 channel type registered for 'Zap'
 [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
 Unable to create channel of type 'Zap' (cause 66 - Channel not
 implemented)
   == Everyone is busy/congested at this time (1:0/0/1)


 My sources are:

 libpri-1.4.1.tar.gz
 zaptel-1.4.5.1.tar.gz
 asterisk-1.4.11.tar.gz
 asterisk-addons-1.4.2.tar.gz
 asterisk-perl-0.10.tar.gz



 I have 1/2 E1  from my provider telephony,  my configuration is

 [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15,17-31
 dchan=16
 loadzone = us
 defaultzone=us





 #cat /etc/asterisk/zapata.conf
 [channels]
 context=default
 switchtype = euroisdn
 pridialplan = unknown
 signalling = pri_cpe
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 immediate=no
 amaflags=documentation
 musiconhold=default


 ;Configure Channels
 group=0
 callgroup=0
 pickupgroup=0
 channel = 1-4
 group=1
 callgroup=1
 pickupgroup=1
 channel = 5-8
 group=2
 callgroup=2
 pickupgroup=2
 channel = 9-12
 group=3
 callgroup=3
 pickupgroup=3
 channel = 13-14
 group=4
 callgroup=4
 pickupgroup=4
 channel = 15


 I have could make calls but, after of some minutes my server is hung,
 suggestions are welcome. Thanks for any help in advance.

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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Brian Hutchinson
I feel more comfortable with MySQL ... just need to learn how to get the
dialplan to use it.

Also figure out the pro's/con's to MySQL vs AstDB.  If I used MySQL then I
could put myphpadmin and get a pseudo GUI to manipulate the blacklist
database for almost no effort so that is another reason for leaning towards
MySQL.

Thanks for the nudge toward MySQL.


On 10/18/07, Lenz [EMAIL PROTECTED] wrote:


 It's not technically complex to do - you can probably use the astdb for
 that, or store all incoming numbers with timestamp in MySQL and run
 something like:

 SELECT count(*)  5 AS blacklisted
 FROM incoming_calls
 WHERE callerid = 12345
 AND timestamp  DATE_SUB( NOW(), INTERVAL 15 MINUTE )

 you should be very well aware of the risks that can stem from such a
 program - in case of bugs, or anomalous situations, you might end up
 blacklisting somebody who actually needs to call in.

 I hope this helps
 l.










 On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson
 [EMAIL PROTECTED] wrote:

  Hi,
 
  I've been reading all I can on Google (and Asterisk TFOT book) looking
  for
  ideas on how to implement an automated blacklist feature.
 
  I would like to automatically blacklist a incoming number based on
  timestamp
  and count information.
 
  For example, if I get a prank call from the same number 5 times within
 15
  minutes, I want my dialplan to automatically blacklist this number.
 
  Should I be looking at AGI to do something like this?
 
  Thanks for any ideas or pointers!



 --
 Loway Research - Home of QueueMetrics
 http://queuemetrics.com

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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Jared Smith
On Thu, 2007-10-18 at 09:34 -0500, Pablo Almido wrote:
 [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
 channel type registered for 'Zap'
 [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
 Unable to create channel of type 'Zap' (cause 66 - Channel not
 implemented)
   == Everyone is busy/congested at this time (1:0/0/1)

That would seem to indicate that the chan_zap.so module isn't being
loaded.  What happens if you type module unload chan_zap.so and then
module load chan_zap.so from the Asterisk CLI?  I'll bet you'll find
that either there's a problem in your zapata.conf file, or that chan_zap
hasn't been compiled for some reason.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Forrest Beck
How do you know that the call is a prank call, an not just someone  
that likes calling your company alot... ?


If you just want a database of callerid's to block, here is what I  
have used, I hope it helps some


My SQL table looks has 4 columns id (autoincrement), callerid,  
blockenabled (to enable or disable the block), and notes.


[general]
realdb_host=localhost
realdb_user=asterisk
realdb_pass=password
realdb_db=asterisk_realtime

[pri-in]
; Conference Room Number
exten = 193,1,Answer()
exten = 193,2,Macro(checkblacklist,${CALLERID(num)})
exten = 193,3,GoTo(us-conference,s,1)

[macro-checkblacklist]
; This Macro will check the blacklist table to see if the callerid of  
the
; caller exist and blockenabled =1 (TRUE). If the callerid is listed,  
then

; tell the caller they have been blacklisted and politely HangUp()
;
; ${ARG1} = CallerID of incoming call
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $ 
{realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\  
blacklist\ where\ callerid=${ARG1} and blockenabled = 1)

exten = s,3,MYSQL(Fetch fetchid ${resultid} blacklistid)
exten = s,4,MYSQL(Clear ${resultid})
exten = s,5,MYSQL(Disconnect ${connid})
exten = s,6,GoToIf($[${blacklistid} = ]?7:fail,1)
exten = s,7,NoOp(Not blocked in Blacklist)
; If the callerid is listed in the database, then send to  
blacklistednumber

;  context
;
exten = fail,1,NoOp(${blacklistid})
exten = fail,2,GoTo(blacklistednumber,s,1)

[blacklistednumber]
; This is where a call will land if the macro-checkblacklist decides  
that

; the number should not be allowed to dial DA
exten = s,1,Wait(2)
exten = s,2,Playback(privacy-you-are-blacklisted)
exten = s,3,HangUp()


Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz



On Oct 18, 2007, at 10:25 AM, Lenz wrote:



It's not technically complex to do - you can probably use the astdb  
for

that, or store all incoming numbers with timestamp in MySQL and run
something like:

SELECT count(*)  5 AS blacklisted
 FROM incoming_calls
WHERE callerid = 12345
AND timestamp  DATE_SUB( NOW(), INTERVAL 15 MINUTE )

you should be very well aware of the risks that can stem from such a
program - in case of bugs, or anomalous situations, you might end up
blacklisting somebody who actually needs to call in.

I hope this helps
l.










On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson
[EMAIL PROTECTED] wrote:


Hi,

I've been reading all I can on Google (and Asterisk TFOT book)  
looking

for
ideas on how to implement an automated blacklist feature.

I would like to automatically blacklist a incoming number based on
timestamp
and count information.

For example, if I get a prank call from the same number 5 times  
within 15

minutes, I want my dialplan to automatically blacklist this number.

Should I be looking at AGI to do something like this?

Thanks for any ideas or pointers!




--
Loway Research - Home of QueueMetrics
http://queuemetrics.com

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Re: [asterisk-users] Issues with Zaptel 1.4.5.1

2007-10-18 Thread Tzafrir Cohen
On Thu, Oct 18, 2007 at 08:35:01AM +0100, Alan Lord wrote:
 Tzafrir Cohen wrote:
 snip /... is required to properly start zaptel. It will also run 
 ztcfg. Otherwise
  users run into issues where misconfigured zaptel.conf fails loading of a
  module. That is a buggy behaviour.
  
  If your card is an analog one, take a look at http://bugs.digium.com/7613
  and tell me what you think.
  
  Something similar for digital spans would require more information in
  sysfs.
  
 
 So do I understand this correctly? I can use this patch to alleviate the 
 need for a userspace init.d start-up script?
 
 I just patch the zaptel module source and rebuild?
 
 Does it read /etc/zaptel.conf and thus load the zone data?
 
 I only have an x100p so this could be a useful patch for me.

As I mentioned in recent notes: that patch is not up-to-date. It also
does not set the tone zones.

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] BBC on Atserix

2007-10-18 Thread Drew Gibson

Tzafrir Cohen wrote:

On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
  

Just for fun.

http://news.bbc.co.uk/1/hi/magazine/7049642.stm



It's Asterix != Asterisk. Though named after *.

  


In Britain, it's called humour :-)

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Lenz

Yeah I would use MySQL as well for more or less the same reasons. Using  
MySQL right from the dialplan is not very elegant but it's pretty simple -  
see http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MYSQL
Thanks
l.



On Thu, 18 Oct 2007 16:49:48 +0200, Brian Hutchinson  
[EMAIL PROTECTED] wrote:

 I feel more comfortable with MySQL ... just need to learn how to get the
 dialplan to use it.

 Also figure out the pro's/con's to MySQL vs AstDB.  If I used MySQL then  
 I
 could put myphpadmin and get a pseudo GUI to manipulate the blacklist
 database for almost no effort so that is another reason for leaning  
 towards
 MySQL.

 Thanks for the nudge toward MySQL.


 On 10/18/07, Lenz [EMAIL PROTECTED] wrote:


 It's not technically complex to do - you can probably use the astdb for
 that, or store all incoming numbers with timestamp in MySQL and run
 something like:

 SELECT count(*)  5 AS blacklisted
 FROM incoming_calls
 WHERE callerid = 12345
 AND timestamp  DATE_SUB( NOW(), INTERVAL 15 MINUTE )

 you should be very well aware of the risks that can stem from such a
 program - in case of bugs, or anomalous situations, you might end up
 blacklisting somebody who actually needs to call in.

 I hope this helps
 l.










 On Thu, 18 Oct 2007 15:02:11 +0200, Brian Hutchinson
 [EMAIL PROTECTED] wrote:

  Hi,
 
  I've been reading all I can on Google (and Asterisk TFOT book) looking
  for
  ideas on how to implement an automated blacklist feature.
 
  I would like to automatically blacklist a incoming number based on
  timestamp
  and count information.
 
  For example, if I get a prank call from the same number 5 times within
 15
  minutes, I want my dialplan to automatically blacklist this number.
 
  Should I be looking at AGI to do something like this?
 
  Thanks for any ideas or pointers!



 --
 Loway Research - Home of QueueMetrics
 http://queuemetrics.com

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-- 
Loway Research - Home of QueueMetrics
http://queuemetrics.com

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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Brian West
Why would a config error stop the module from loading? That seems  
like a suboptimal behavior.


/b

On Oct 18, 2007, at 9:50 AM, Jared Smith wrote:


That would seem to indicate that the chan_zap.so module isn't being
loaded.  What happens if you type module unload chan_zap.so and then
module load chan_zap.so from the Asterisk CLI?  I'll bet you'll find
that either there's a problem in your zapata.conf file, or that  
chan_zap

hasn't been compiled for some reason.


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Re: [asterisk-users] Is anyone successfully using IMAP storage

2007-10-18 Thread Jared Bellows
On 10/18/07, Olivier [EMAIL PROTECTED] wrote:

 Hi,

 From personal experience, Asterisk 1.4 IMAP storage seems broken and
 unusable.
 Is anyone using it successfully ?


I've read blog entries that indicate that people have used it successfully
but I have not been able to get it to connect to the IMAP server in all my
trials.
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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Pablo Almido
Yes, the module is load

# asterisk  -r

ippbx*CLI module show like chan_zap.so
Module Description
 Use Count
chan_zap.soZapata Telephony
 0
1 modules loaded
ippbx*CLI
ippbx*CLI




2007/10/18, Brian West [EMAIL PROTECTED]:
 Make sure chan_zap.so is loaded.

 /b

 On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote:

  Hi List,
 
  I am from Peru, I have installed an asterisk server in my company with
  digium card E1 TE120P, I am having issues when i make calls, here the
  error from my server
 
 
  [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
  channel type registered for 'Zap'
  [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
  Unable to create channel of type 'Zap' (cause 66 - Channel not
  implemented)
== Everyone is busy/congested at this time (1:0/0/1)
 
 
  My sources are:
 
  libpri-1.4.1.tar.gz
  zaptel-1.4.5.1.tar.gz
  asterisk-1.4.11.tar.gz
  asterisk-addons-1.4.2.tar.gz
  asterisk-perl-0.10.tar.gz
 
 
 
  I have 1/2 E1  from my provider telephony,  my configuration is
 
  [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
  span=1,1,0,ccs,hdb3,crc4
  bchan=1-15,17-31
  dchan=16
  loadzone = us
  defaultzone=us
 
 
 
 
 
  #cat /etc/asterisk/zapata.conf
  [channels]
  context=default
  switchtype = euroisdn
  pridialplan = unknown
  signalling = pri_cpe
  usecallerid=yes
  hidecallerid=no
  callwaiting=yes
  callwaitingcallerid=yes
  threewaycalling=yes
  transfer=yes
  canpark=yes
  cancallforward=yes
  callreturn=yes
  echocancel=yes
  echocancelwhenbridged=yes
  rxgain=0.0
  txgain=0.0
  immediate=no
  amaflags=documentation
  musiconhold=default
 
 
  ;Configure Channels
  group=0
  callgroup=0
  pickupgroup=0
  channel = 1-4
  group=1
  callgroup=1
  pickupgroup=1
  channel = 5-8
  group=2
  callgroup=2
  pickupgroup=2
  channel = 9-12
  group=3
  callgroup=3
  pickupgroup=3
  channel = 13-14
  group=4
  callgroup=4
  pickupgroup=4
  channel = 15
 
 
  I have could make calls but, after of some minutes my server is hung,
  suggestions are welcome. Thanks for any help in advance.
 
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[asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Julian Lyndon-Smith
Apart from religious grounds (!), is there any pros or cons why I should 
choose one over the other for a new install of asterisk ?

Julian

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Re: [asterisk-users] Is anyone successfully using IMAP storage

2007-10-18 Thread [EMAIL PROTECTED]
On 10/18/07, Olivier [EMAIL PROTECTED] wrote:
 Hi,

 From personal experience, Asterisk 1.4 IMAP storage seems broken and
 unusable.
 Is anyone using it successfully ?

 This kind of poll would be very useful to estimate is a code rewrite has a
 chance to disturb a running system.
 I we get no successful report, it would help developers to consider a code
 rewrite as patch instead of a new feature.

 So please, do not hesitate to report successful or unsuccessful or never
 tried use of IMAP storage, it will increase our chances to get this feature
 in 1.4 branch and not wait for 1.6.

 Cheers

I wanted to use it but there's *NO* documentation. We don't use
dovecot but our IMAP server has a master user function... can't figure
out if its the same as the dovecot function, because:

1) The dovecot function isn't documented
2) There's no documentation as to how asterisk writes to the IMAP.

So I'll put it out to the world to see if someone knows how it
works Our email server can do this:

You can log into any mailbox via IMAP if your user has mboxadmin
capabilities. So if you want to use UserA to log into UserB's mailbox,
you can do the following:

Username: mboxadmin:UserA:UserB
Password: Password_User_A

Is that what Asterisk is expecting?

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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Pablo Almido
I have unload and load the module, it is output


ippbx*CLI module unload chan_zap.so
  == Unregistered application 'ZapSendKeypadFacility'

ippbx*CLI module load chan_zap.so
  == Registered application 'ZapSendKeypadFacility'
  == Parsing '/etc/asterisk/zapata.conf': Found
[Oct 18 10:46:38] WARNING[2790]: chan_zap.c:903 zt_open: Unable to
specify channel 1: No such device or address
[Oct 18 10:46:38] ERROR[2790]: chan_zap.c:7160 mkintf: Unable to open
channel 1: No such device or address
here = 0, tmp-channel = 1, channel = 1
[Oct 18 10:46:38] ERROR[2790]: chan_zap.c:10466 build_channels: Unable
to register channel '1-4'

ippbx*CLI





2007/10/18, Brian West [EMAIL PROTECTED]:
 Why would a config error stop the module from loading? That seems like a
 suboptimal behavior.

 /b


 On Oct 18, 2007, at 9:50 AM, Jared Smith wrote:


 That would seem to indicate that the chan_zap.so module isn't being

 loaded.  What happens if you type module unload chan_zap.so and then

 module load chan_zap.so from the Asterisk CLI?  I'll bet you'll find

 that either there's a problem in your zapata.conf file, or that chan_zap

 hasn't been compiled for some reason.

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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Tzafrir Cohen
On Thu, Oct 18, 2007 at 12:22:24PM -0400, [EMAIL PROTECTED] wrote:

 
 Just 5 months ago CENTOS started to use Linux 2.6 

Centos 4 (based on RHEL4) used kernel 2.6 as well.
It was released over two years ago (it has 2.6.9).

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Brian West
I'm sorry I call bullshit on this one.  CentOS has been 2.6 for some  
time.


/b

On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote:


Just 5 months ago CENTOS started to use Linux 2.6 one of the
reasons I'd abandoned for SuSE a while back.


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Per Jessen
Julian Lyndon-Smith wrote:

 Apart from religious grounds (!), is there any pros or cons why I
 should choose one over the other for a new install of asterisk ?
 

I doubt it.  A distro is a distro. 


/Per Jessen, Zürich
We use only openSUSE.


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Re: [asterisk-users] Anyone having any luck with Bluetooth?

2007-10-18 Thread Steve Murphy
On Wed, 2007-10-17 at 15:09 -0700, shadowym wrote:

 I have read all the wiki's and blogs and how to links about Bluetooth but so
 far no luck.  I can confirm that CentOS5 sees my Bluetooth adapter and my
 cell phone.  No Joy on Asterisk 1.4.  The information out there is kind of
 confusing as there is a lot of outdated info sometimes referring to software
 no longer actively developed.
 
 What I think I have managed to conclude is that there is a Bluetooth module
 for Asterisk 1.4 that supposedly works but it's not part of any released
 branches so I will have to use a development branch. That is ok but I still
 can't get it to work.  
 
 I want to use my Cell phone as a secondary trunk for a business so I need it
 to work reasonably well.  Does it (if and when I get it working)?  Is there
 a good (up to date relevant) how to somewhere?   
 
 


Haven't tried chan_mobile against 1.4; just against trunk; and have it
working
OK with 2 of 3 bluetooth dongles with my samsung cellphone.

Look in asterisk-addons/trunk for chan_mobile.c; set up your config
file; mine looks like this:

;
; configuration file for chan_mobile
;

[general]
interval=30 ; Number of seconds between trying to connect to 
devices. 

; The following is a list of adapters we use.
; id must be unique and address is the bdaddr of the adapter from
hciconfig.
; Each adapter may only have one device (headset or phone) connected at
a time.
; Add an [adapter] entry for each adapter you have.

[adapter]
id=encore
address=00:11:C6:09:DF:99

[adapter]
id=iogear_gbu221
address=00:02:49:07:FC:44

[adapter]
id=asus
address=00:02:72:27:92:8D

; The following is a list of the devices we deal with.
; Every device listed below will be available for calls in and out of
Asterisk. 
; Each device needs an adapter= entry which determines which
bluetooth adapter is used.
; Use the CLI command 'mobile search' to discover devices.
; Use the CLI command 'mobile show devices' to see device status.
;
; To place a call out through a mobile phone use
Dial(Mobile/[device]/NNN.) or Dial(Mobile/gn/NNN..) in your
dialplan.
; To call a headset use Dial(Mobile/[device]).


[steve]
context=incoming
address=00:1B:62:52:F4:F2
port=4
adapter=encore
group=1


;; The following is a list of the devices we deal with.
;; Every device listed below will be available for calls in and out of
Asterisk. 
;; Discovered devices not in this list are not available.
;; Use the CLI command 'mobile search' to discover devices.
;; Use the CLI command 'mobile show devices' to see device status.
;;
;; To place out through a cell phone use Dial(Mobile/[device]/NNN.)
in your dialplan.
;; To call a headset use Dial(Mobile/[device]).
;;


-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread [EMAIL PROTECTED]
On 10/18/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
 Apart from religious grounds (!), is there any pros or cons why I should
 choose one over the other for a new install of asterisk ?

 Julian

SuSE is known for using the latest packages with each release, and
RHEL/CENTOS are known for using old citing stability. SuSE Is just
as stable, they don't use beta version only tested stable releases.

Just 5 months ago CENTOS started to use Linux 2.6 one of the
reasons I'd abandoned for SuSE a while back.

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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Tilghman Lesher
On Thursday 18 October 2007 09:49:48 Brian Hutchinson wrote:
 I feel more comfortable with MySQL ... just need to learn how to get the
 dialplan to use it.

See func_odbc.conf.

-- 
Tilghman

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Re: [asterisk-users] Softphone that emulates Skype API ?

2007-10-18 Thread Tzafrir Cohen
On Thu, Oct 18, 2007 at 01:05:16PM +0300, Cosmin Prund wrote:
 There's a large number of gadgets one can buy that work with Skype
 through the API. One of the things I'm interested right now is the
 ability to properly use a mobile phone headset with a SIP/IAX softphone.
 
  
 
 Is there an softphone that emulates the Skype API?
 
 Are there legal implications in writing an softphone that emulates the
 Skype API?
 
 Should I just give up and buy a Siemens DECT phone that supports a
 bluetooth headset?

Many of the skype phones can at least be used as an extra sound
device. Using them as a handset is probably more complicated.

-- 
   Tzafrir Cohen   
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Tzafrir Cohen
On Thu, Oct 18, 2007 at 10:53:15AM -0500, Pablo Almido wrote:
 I have unload and load the module, it is output
 
 
 ippbx*CLI module unload chan_zap.so
   == Unregistered application 'ZapSendKeypadFacility'
 
 ippbx*CLI module load chan_zap.so
   == Registered application 'ZapSendKeypadFacility'
   == Parsing '/etc/asterisk/zapata.conf': Found
 [Oct 18 10:46:38] WARNING[2790]: chan_zap.c:903 zt_open: Unable to
 specify channel 1: No such device or address
 [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:7160 mkintf: Unable to open
 channel 1: No such device or address
 here = 0, tmp-channel = 1, channel = 1
 [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:10466 build_channels: Unable
 to register channel '1-4'

cat /proc/zaptel/1

Did it get properly configured?

http://rapid.tzafrir.org.il/docs/README.html#toc20 (the section about
the procfs interface)

-- 
   Tzafrir Cohen   
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread [EMAIL PROTECTED]
On 10/18/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Thu, Oct 18, 2007 at 12:22:24PM -0400, [EMAIL PROTECTED] wrote:

 
  Just 5 months ago CENTOS started to use Linux 2.6

 Centos 4 (based on RHEL4) used kernel 2.6 as well.
 It was released over two years ago (it has 2.6.9).

Never trust anyone... I've been anti-Cent/RHEL for a while because of
these guys:

http://distrowatch.com/table.php?distribution=centos

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Re: [asterisk-users] Blocking collect calls in Brazil

2007-10-18 Thread Luis Antonio Prata Barbosa
Hi Carlos,

The method commonly used is flash E1 to block this calls.
A Better way could be detect the category of call.
I don´t know if there is a way to get the call category in extensions as we
can get the CALLERID.

Collect calls have B-8 category in Brazil.

Thanks.

Luis A P Barbosa

2007/10/18, Carlos Barros [EMAIL PROTECTED]:

 Hi list.

 I was googling about this subject and found a message to this list
 asking the same thing I want, but see no response. So, I'm reposting
 it in case someone has a solution for this.
 I'd like to know if there's some way to detect (not block) a collect
 call in a ISDN E1? I need this cause I have my own billing system and
 I need to charge my customers for collect calls they receive.
 Currently I have to get the PSTN billing and locate all collect calls
 and then enter it manually. I want to detect these calls in asterisk
 so I can automatically charge my customers.

 For Brazilians in the list, I have a Embratel E1 - ISDN.

 Thanks

 Carlos Barros

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[asterisk-users] Incoming calls

2007-10-18 Thread Gustavo Gonzalez
Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card and
when I get a free Zap channel to call I hear the voice of a people on the
other side from an incomming call, I think that asterisk bridge my free
channel with incomming calls but how do this?Thanks for any idea.

Alejandro González
Grupo Gestión
4384-0660
www.grupo-gestion.com.ar
[EMAIL PROTECTED]
---

---
RI 9000-1069
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Certificado por IRAM
Norma ISO: 9001-2000
 
 
 
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Re: [asterisk-users] BBC on Atserix

2007-10-18 Thread SIP
Drew Gibson wrote:
 Tzafrir Cohen wrote:
 On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
   
 Just for fun.

 http://news.bbc.co.uk/1/hi/magazine/7049642.stm
 

 It's Asterix != Asterisk. Though named after *.

   

 In Britain, it's called humour :-)

 regards,

 Drew


Only if it's actually humourous. ;)

N.

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Re: [asterisk-users] Automating blacklists

2007-10-18 Thread Steve Murphy
On Thu, 2007-10-18 at 16:02 +0300, Brian Hutchinson wrote:
 Hi,
 
 I've been reading all I can on Google (and Asterisk TFOT book) looking
 for ideas on how to implement an automated blacklist feature.
 
 I would like to automatically blacklist a incoming number based on
 timestamp and count information. 
 
 For example, if I get a prank call from the same number 5 times within
 15 minutes, I want my dialplan to automatically blacklist this number.
 
 Should I be looking at AGI to do something like this?
 
 Thanks for any ideas or pointers! 

I hate telemarketers.

And at ***least*** one calls every day.

But I rarely have to talk to them.

Using AstDB, and dialplan programming in AEL, I have implemented some
measures
that cut the number of telemarketers who make it through to only maybe
1 per year.

All my stuff is in AEL. Along this line, my home phone system records
all CID's coming in, and counts each incoming call for each CID.
Telemarketer autodialers tend to call in without any CID. So, I play
anonymous callers the telezapper tri-tones. It's kinda irritating to
callers to get this, so I only play it to eligible callers. Autodialers
usually immediately hang up, and because they usually interpret
tri-tones as wrong number, they remove me from their db, and I don't
hear from them again... But there is a growing number of telemarketers
who actually supply some sort of CID info for the call. I count how many
times they call, but get filtered out. After I get the same party
calling in 4 times or more in row, that don't actually end up talking to
anyone, I start playing them the tri-tones. That usually is the end of
that sequence.

All this is done with simple dialplan statements and the use of AstDB
(see the DB() function).

I also whitelist, and route certain callers directly to particular
extensions, bypassing all filters.

I also have DB entries that will override the default MOH for some
callers, based either on CID, or who they are calling. For instance, one
of my sons preferred his callers got rock music instead of the normal
elevator stuff.

Also available to those who prefer not to be bothered continuously by
charity seekers, political pollsters, long-distance dialing hawkers,
septic-tank bacteria salesmen, etc, are the privacy related options to
the Dial() app. You can do call screening, and route unwanted incoming
calls to voicemail. Might even help with debt collectors, etc.


murf



-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] Multi-site / Multi-server coordination

2007-10-18 Thread Daniel Hazelbaker
Okay, so we are planning for the future here where I work so we are  
trying to do testing ahead of time as we might be setting up a  
satellite campus that would need its own Asterisk phone system but  
still tied into our main campus phone system.  This much we have  
accomplished.  We have a central database that routes calls to the  
correct server before dialing the final destination SIP phone (or  
Zap, IAX, whatever).

Our last big question is this: How, if at all, does asterisk deal  
with multi-server voicemail?  i.e. User A is at site A and has a  
voicemail in his mailbox.  He knows it really belongs to user B (who  
is at site B) so after listening to the voicemail he forwards it to  
user B.  Is there a way for Asterisk to realize this box belongs on a  
different machine and communicate the voicemail over the network to  
that machine, or is this something I need to write? :)

Daniel Hazelbaker
Information Technology Director
High Desert Church

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Re: [asterisk-users] Crash related to asterisk -rx ?

2007-10-18 Thread Jean-Denis Girard
Atis Lezdins a écrit :

 Yup, it's also a problem for me, but it haven't ever crashed server. It just 
 makes specific remote process unresponsive. There's a patch for 1.4, but i 
 guess it wouldn't be hard to backport it for 1.2
 
 http://bugs.digium.com/view.php?id=10847
 
 you might also want the one mentioned in comments:
 
 http://bugs.digium.com/view.php?id=10888
 
 Regards,
 Atis
 

Atis,

Thanks for the reply and pointers.

Best regards,
-- 
Jean-Denis Girard

SysNux  Systèmes Linux en Polynésie française
http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527

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Re: [asterisk-users] BBC on Atserix

2007-10-18 Thread Drew Gibson

SIP wrote:

Drew Gibson wrote:
  

Tzafrir Cohen wrote:


On Thu, Oct 18, 2007 at 01:04:06PM +0100, Cartwright, Dave wrote:
  
  

Just for fun.

http://news.bbc.co.uk/1/hi/magazine/7049642.stm



It's Asterix != Asterisk. Though named after *.

  
  

In Britain, it's called humour :-)

regards,

Drew




Only if it's actually humourous. ;)

N.

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It's looking good ...

http://news.bbc.co.uk/2/hi/entertainment/4343264.stm

regards,

Drew


--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Jay R. Ashworth
On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote:
 Julian Lyndon-Smith wrote:
  Apart from religious grounds (!), is there any pros or cons why I
  should choose one over the other for a new install of asterisk ?
 
 I doubt it.  A distro is a distro. 

Well, no.

 /Per Jessen, Zürich
 We use only openSUSE.

If A distro is a distro, then why?

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Pablo Almido
I run this command

[EMAIL PROTECTED] ~]# cat /proc/zaptel/1
Span 1: WCT1/0 Wildcard TE12xP Card 0
IRQ misses: 40

   1 WCT1/0/1
   2 WCT1/0/2
   3 WCT1/0/3
   4 WCT1/0/4
   5 WCT1/0/5
   6 WCT1/0/6
   7 WCT1/0/7
   8 WCT1/0/8
   9 WCT1/0/9
  10 WCT1/0/10
  11 WCT1/0/11
  12 WCT1/0/12
  13 WCT1/0/13
  14 WCT1/0/14
  15 WCT1/0/15
  16 WCT1/0/16
  17 WCT1/0/17
  18 WCT1/0/18
  19 WCT1/0/19
  20 WCT1/0/20
  21 WCT1/0/21
  22 WCT1/0/22
  23 WCT1/0/23
  24 WCT1/0/24
  25 WCT1/0/25
  26 WCT1/0/26
  27 WCT1/0/27
  28 WCT1/0/28
  29 WCT1/0/29
  30 WCT1/0/30
  31 WCT1/0/31


Then I run ztcfg

[EMAIL PROTECTED] ~]# ztcfg -vv

Zaptel Version: 1.4.5.1
Echo Canceller: MG2
Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) (Slaves: 14)
Channel 15: Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Clear channel (Default) (Slaves: 17)
Channel 18: Clear channel (Default) (Slaves: 18)
Channel 19: Clear channel (Default) (Slaves: 19)
Channel 20: Clear channel (Default) (Slaves: 20)
Channel 21: Clear channel (Default) (Slaves: 21)
Channel 22: Clear channel (Default) (Slaves: 22)
Channel 23: Clear channel (Default) (Slaves: 23)
Channel 24: Clear channel (Default) (Slaves: 24)
Channel 25: Clear channel (Default) (Slaves: 25)
Channel 26: Clear channel (Default) (Slaves: 26)
Channel 27: Clear channel (Default) (Slaves: 27)
Channel 28: Clear channel (Default) (Slaves: 28)
Channel 29: Clear channel (Default) (Slaves: 29)
Channel 30: Clear channel (Default) (Slaves: 30)
Channel 31: Clear channel (Default) (Slaves: 31)

31 channels configured.

Changing signalling on channel 1 from Unused to Clear channel
Changing signalling on channel 2 from Unused to Clear channel
Changing signalling on channel 3 from Unused to Clear channel
Changing signalling on channel 4 from Unused to Clear channel
Changing signalling on channel 5 from Unused to Clear channel
Changing signalling on channel 6 from Unused to Clear channel
Changing signalling on channel 7 from Unused to Clear channel
Changing signalling on channel 8 from Unused to Clear channel
Changing signalling on channel 9 from Unused to Clear channel
Changing signalling on channel 10 from Unused to Clear channel
Changing signalling on channel 11 from Unused to Clear channel
Changing signalling on channel 12 from Unused to Clear channel
Changing signalling on channel 13 from Unused to Clear channel
Changing signalling on channel 14 from Unused to Clear channel
Changing signalling on channel 15 from Unused to Clear channel
Changing signalling on channel 16 from Unused to HDLC with FCS check
Changing signalling on channel 17 from Unused to Clear channel
Changing signalling on channel 18 from Unused to Clear channel
Changing signalling on channel 19 from Unused to Clear channel
Changing signalling on channel 20 from Unused to Clear channel
Changing signalling on channel 21 from Unused to Clear channel
Changing signalling on channel 22 from Unused to Clear channel
Changing signalling on channel 23 from Unused to Clear channel
Changing signalling on channel 24 from Unused to Clear channel
Changing signalling on channel 25 from Unused to Clear channel
Changing signalling on channel 26 from Unused to Clear channel
Changing signalling on channel 27 from Unused to Clear channel
Changing signalling on channel 28 from Unused to Clear channel
Changing signalling on channel 29 from Unused to Clear channel
Changing signalling on channel 30 from Unused to Clear channel
Changing signalling on channel 31 from Unused to Clear channel




Then I back to run this command , but I can not understand why it
change in channel 16, However I can not  make calls


[EMAIL PROTECTED] ~]# cat /proc/zaptel/1
Span 1: WCT1/0 Wildcard TE12xP Card 0 HDB3/CCS/CRC4
IRQ misses: 40

   1 WCT1/0/1 Clear
   2 WCT1/0/2 Clear
   3 WCT1/0/3 Clear
   4 WCT1/0/4 Clear
   5 WCT1/0/5 Clear
   6 WCT1/0/6 Clear
   7 WCT1/0/7 Clear
   8 WCT1/0/8 Clear

Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Per Jessen
Jay R. Ashworth wrote:

 On Thu, Oct 18, 2007 at 06:25:39PM +0200, Per Jessen wrote:
 Julian Lyndon-Smith wrote:
  Apart from religious grounds (!), is there any pros or cons why I
  should choose one over the other for a new install of asterisk ?
 
 I doubt it.  A distro is a distro.
 
 Well, no.
 
 /Per Jessen, Zürich
 We use only openSUSE.
 
 If A distro is a distro, then why?

Because we like it, and because we're used to it.  That we have an
operational preference doesn't change a distro is a distro.  


/Per Jessen, Zürich

-- 
http://www.spamchek.com/ - your spam is our business.


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Perssy Llamosas
I doubt it.

hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/

 Original Message 
Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 18/10/2007 11:25 a.m.
 Julian Lyndon-Smith wrote:

   
 Apart from religious grounds (!), is there any pros or cons why I
 should choose one over the other for a new install of asterisk ?

 

 I doubt it.  A distro is a distro. 


 /Per Jessen, Zürich
 We use only openSUSE.


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Tzafrir Cohen
On Thu, Oct 18, 2007 at 02:54:36PM -0500, Perssy Llamosas wrote:
 I doubt it.
 
 hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/

The above page (even with the proper protocol name) is very low on
actual facts and reasonings. It mentions nothing specific to openSUSE
10.3 .

So let's get back to the topic:
anybody tried the package of Asterisk included in 10.3?

-- 
   Tzafrir Cohen   
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Incoming calls

2007-10-18 Thread C F
Glare that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.

http://www.telos-systems.com/techtalk/gldefs.htm#Glare


On 10/18/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote:
 Hello I have a question about incoming calls on TDM400P cards. I want to
 know why an incoming call appear in a sorpresive way on a phone that I
 pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
 channels )that are free. I have four lines connected to my TDM400P card and
 when I get a free Zap channel to call I hear the voice of a people on the
 other side from an incomming call, I think that asterisk bridge my free
 channel with incomming calls but how do this?Thanks for any idea.

 Alejandro González
 Grupo Gestión
 4384-0660
 www.grupo-gestion.com.ar
 [EMAIL PROTECTED]
 ---

 ---
 RI 9000-1069
 Sistema de Gestión de Calidad
 Certificado por IRAM
 Norma ISO: 9001-2000



 --
 No virus found in this outgoing message.
 Checked by AVG Free Edition.
 Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006
 01:45 p.m.


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[asterisk-users] Ring Groups

2007-10-18 Thread Rob Schall
Here's what I'm looking to do

exten = 10,1,Dial(SIP/1000SIP/1001,15,wW)
exten = 10,2,Voicemail(u1000)


This should ring both phones and they should keep ringing for the
alloted time before moving on. However, it appears that if one of the
phones is Busy, it returns with a busy and moves on without really
ringing the second phone.

Short of checking if the channels are available or using a queue, is
there a way to ignore the return value and just make it ring for 10
seconds and then move on to the second step?

Any Suggestions?


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[asterisk-users] First Time T1 Questions

2007-10-18 Thread Michael J. Liberatore
Hi all, i have been using asterisk for a few years but i am about to do
my first t1 setup.  After terrible quality issues between two business
locations, we have decided to purchase a point to point t1 from the
local phone co.  The internet is too crappy, too much lag, queing and
jitter.  Most calls were dropped.
 
I was about to order two cisco routers with csu cards and remembered our
wonderful asterisk supports direct t1.  I remembered digium and sangoma
both make these cards.
 
After some problems with a digium fxo card, i just ordered a sangoma
a200 with echo cancellation.  I was also leaning towards getting the
single t1 sangoma card that is $499 from voip supply.  But i know digium
also makes one.  I was wondering if the digium card works better or much
easier with asterisk?  The digium description says you can split the t1
for voice and data which sounds nice since i will only be using probably
4 channels max of the t1.  Does the sangoma card also do this?  I
noticed the sangoma card has a 5 year warranty which is nice since i
have had multiple digium fxo cards die.  Is there any other reason to
get or the other?  
 
Thank you all for your help.  I am hoping this opens up a whole new
world in asterisk for me.
 
-Mike
 
 


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Re: [asterisk-users] 64 bit asterisk

2007-10-18 Thread Baji Panchumarti
 I hope you have better success than I did, my problem was
 not so much with asterisk in particular but 64-bit in general.

 Examples of problems using CentOS 4.5 on x86_64

 - many problems loading php5  mysql from package
   repositories.

 - a few asterisk functions don't work, eg  STRFTIME()

 Perhaps the distro you are using is more caught up on
 64 bit.

 Everything upgraded/updated without a hitch on 32 bit.

 64 bit is a no go unless you are running packages that
 have matured for atleast a couple of years old...imho.

--

 On 10/18/07, joakimsen wrote:

 Nope it should just work. Just finished setting up 1.4 for the first
 time in a while and just works. Been running 1.2 for the longest time
 and same thing.

 On 10/18/07, Wai Wu wrote:
  Hi list,
 
  I just installed 64 bit Linux, and ready to install Asterisk through
  source on it. Are there any settings have to change to build 64 bit
  Asterisk? Thnx a million.
 
 
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Re: [asterisk-users] Blocking collect calls in Brazil

2007-10-18 Thread Carlos Barros
Hi Luis.

On 10/18/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote:
 Hi Carlos,

 The method commonly used is flash E1 to block this calls.
Yes.. But I need to detect it so I can set a billing on it.
 A Better way could be detect the category of call.
 I don´t know if there is a way to get the call category in extensions as we
 can get the CALLERID.

 Collect calls have B-8 category in Brazil.
Hmm.. I'll investigate it. The callerID does not specify (I guess
Embratel should send a 9 or something as a CAllerID prefix, but
unfortunately not).

Thanks for you information.

Carlos

 Thanks.

 Luis A P Barbosa

 2007/10/18, Carlos Barros [EMAIL PROTECTED]:
 
  Hi list.
 
  I was googling about this subject and found a message to this list
  asking the same thing I want, but see no response. So, I'm reposting
  it in case someone has a solution for this.
  I'd like to know if there's some way to detect (not block) a collect
  call in a ISDN E1? I need this cause I have my own billing system and
  I need to charge my customers for collect calls they receive.
  Currently I have to get the PSTN billing and locate all collect calls
  and then enter it manually. I want to detect these calls in asterisk
  so I can automatically charge my customers.
 
  For Brazilians in the list, I have a Embratel E1 - ISDN.
 
  Thanks
 
  Carlos Barros
 
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[asterisk-users] Free help

2007-10-18 Thread Rony Ron
Hello all,
i would like to have references so i'm giving free help
for any project (commercial or public).

regards,
-- 
Your next Partner !
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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread [EMAIL PROTECTED]
ZAP/G1 will dial starting from channel 1. ZAP/R1 will dial starting
from the last channel of the group.

Pablo, please tell us what version of Linux and which distribution are
you using. Maybe for the time being try the stock asterisk of your
distro or the one they provide in the buildservice?

On 10/18/07, Brett Crapser [EMAIL PROTECTED] wrote:

 Pablo - You said you have 1/2 E1 - which half???
 That might be your problem.  Unless 1/2 E1 means something else...
 Asterisk normally dials out on the low end unless you specify
 G instead of g ??? or something like that.

 Brett


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Re: [asterisk-users] Softphone that emulates Skype API ?

2007-10-18 Thread [EMAIL PROTECTED]
On 10/18/07, Cosmin Prund [EMAIL PROTECTED] wrote:

 One of the things I'm interested right now is the ability to
 properly use a mobile phone headset with a SIP/IAX softphone.


Thats a function of the bluetooth stack. Go ahead and pair your
bluetooth headset to your PC it will work like any audio device. Any
softphone that supports selecting the input source should work. With
the Widcomm bluetooth stack anyways

 Is there an softphone that emulates the Skype API?

Not that I am aware. Skype is proprietary and closed platform.

 Are there legal implications in writing an softphone that emulates the Skype
 API?

Not that I am aware. Even the draconian DMCA laws permit circumvention
of these digital copyright protection mechanisms for software
interoperability.


 Should I just give up and buy a Siemens DECT phone that supports a bluetooth
 headset?


Giving up is your prerogative

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[asterisk-users] 64 bit asterisk

2007-10-18 Thread Wai Wu
Hi list,

I just installed 64 bit Linux, and ready to install Asterisk through
source on it. Are there any settings have to change to build 64 bit
Asterisk? Thnx a million. 


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Perssy Llamosas


I doubt it.

boycottnovell.com/2007/10/02/opensuse-103-release/

 Original Message 
Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 18/10/2007 11:25 a.m.
 Julian Lyndon-Smith wrote:

   
 Apart from religious grounds (!), is there any pros or cons why I
 should choose one over the other for a new install of asterisk ?

 

 I doubt it.  A distro is a distro. 


 /Per Jessen, Zürich
 We use only openSUSE.


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Perssy Llamosas


I doubt it.

boycottnovell.com/2007/10/02/opensuse-103-release

 Original Message 
Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: 18/10/2007 11:25 a.m.
 Julian Lyndon-Smith wrote:

   
 Apart from religious grounds (!), is there any pros or cons why I
 should choose one over the other for a new install of asterisk ?

 

 I doubt it.  A distro is a distro. 


 /Per Jessen, Zürich
 We use only openSUSE.


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Re: [asterisk-users] Issues with making calls

2007-10-18 Thread Brett Crapser

Pablo - You said you have 1/2 E1 - which half???
That might be your problem.  Unless 1/2 E1 means something else...
Asterisk normally dials out on the low end unless you specify
G instead of g ??? or something like that.

Brett


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Dovid B
While I am a fan of CentOS some pople just take it tooo far.

- Original Message - 
From: Perssy Llamosas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, October 18, 2007 9:54 PM
Subject: Re: [asterisk-users] centos 5 vs OpenSuse 10.3


I doubt it.

 hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/

  Original Message 
 Subject: Re:[asterisk-users] centos 5 vs OpenSuse 10.3
 From: Per Jessen [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Date: 18/10/2007 11:25 a.m.
 Julian Lyndon-Smith wrote:


 Apart from religious grounds (!), is there any pros or cons why I
 should choose one over the other for a new install of asterisk ?



 I doubt it.  A distro is a distro.


 /Per Jessen, Zürich
 We use only openSUSE.


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Re: [asterisk-users] centos 5 vs OpenSuse 10.3

2007-10-18 Thread Joel Hill
We used to use CentOS 4 here but about 6-8 months ago we found that
they were too slow with updates  their repos for some of the 3rd party
software that we were developing. We switched to SuSe 10.2 and haven't
looked back. However Asterisk works equally well on both. Just pick your
favorite flavor.

Cheers,

Joel.

On Thu, 2007-10-18 at 11:34 -0500, Brian West wrote:
 I'm sorry I call bullshit on this one.  CentOS has been 2.6 for some
 time.
 
 
 /b
 
 On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote:
 
  Just 5 months ago CENTOS started to use Linux 2.6 one of the
  
  reasons I'd abandoned for SuSE a while back.
  
 
 
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Re: [asterisk-users] ABE 1.4 release date

2007-10-18 Thread Kevin P. Fleming
[EMAIL PROTECTED] wrote:

 Do you think there might be a working t.38 implementation in ABE 1.4?
 Even passthru Between endpoints and gateways with Asterisk in the
 middle?

ABE does not contain functionality that isn't in an open source version
of Asterisk, except for license- or patent-encumbered bits. The T.38
support in ABE version C.x is identical to that in Asterisk 1.4.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] A linksys SPA921 behind NAT and firewall

2007-10-18 Thread [EMAIL PROTECTED]
Did you set NAT Keep Alive Enable: = Yes for the line in question in
the SPA's configuration?



On 10/18/07, Per Jessen [EMAIL PROTECTED] wrote:
 I've got someone sat in a home-office with an SPA921 behind NAT, and
 most probably a firewall.  I've got a STUN-server running, and calling
 in from the SPA921 to our Asterisk box works fine - though I had to
 open the firewall for UDP traffic on port 1-2.

 Calling from our Asterisk to the SPA921 doesn't work.  I'm guessing this
 is due to the NAT/firewall on the other side, coz' how would it know
 that UDP-traffic to SPA publicIP:5060 needs to be delivered to
 192.168.x.x:5060 ?



 /Per Jessen, Zürich

 --
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[asterisk-users] polycom ip330/ip501 second ethernet port

2007-10-18 Thread Robert McNaught
Hi,

Has anyone had any great difficulties with QoS using the second ethernet
phone in these Polycom phones for desktop machines in a converged
network?  I had heard that these can cause difficulties when used in
this manner.  I have always tried to persuade customers to go with 2
ethernet drops per workstation to avoid having to use the phone as a
switch.

I apologize for this question not being directly related to asterisk,
but since Polycom phones are used a lot with asterisk, it seems a good
place to post ;-)

Robert McNaught
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Re: [asterisk-users] Looking for free DID with IAX

2007-10-18 Thread Dovid B



I know I can get free DID's with SIP, is anyone giving out free DID's with
 IAX?

 Thanks in advance,

In the words of the great jbot in the #asterisk channel on irc.freenode.net

Dovid ~ygwypf

jbot well, ygwypf is You Get What You Pay For. If the sole factor in your 
decision to purchase a product or service is that it's cheaper than 
everything else out there, don't be surprised if it's also worse in every 
other respect than everything else out there.



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[asterisk-users] Zaptel 1.2.21 and 1.4.6 released

2007-10-18 Thread Asterisk Development Team
The Asterisk.org development team has announced the release of Zaptel 
versions 1.2.21 and 1.4.6. These releases contain many bug fixes as well 
as performance enhancements (Too many to list here).  A couple of major 
changes: there is an update to the Octasic API version as well as a 
considerable rewrite of the wct4xxp driver.  The xpp drivers have been 
updated quite a bit as well.  There was also a fix for the wctdm24xxp 
driver which sometimes reported false power alarms.  For further details 
as well as the additional changes, see the respective Changelogs.

Both releases are available as a tarball as well as a patch against the 
previous release. They are available for download from downloads.digium.com.

Thank you for your support!

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Re: [asterisk-users] Limit number of times a call can be forwarded

2007-10-18 Thread Dovid B

- Original Message - 
From: Don Pobanz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, October 18, 2007 3:38 PM
Subject: [asterisk-users] Limit number of times a call can be forwarded


 We have had a few different times when a user has forwarded their phone
 to himself. This has overloaded the communications to our operator panel
 (FOP). One user should not be able to effect the whole phone system!

 Is there a way that the number of times that a call can be forwarded
 could be limited like to 10 or even 100? Then even if a user does
 something stupid like forwarding their calls to himself, it wouldn't
 cause problems for others.

 Don Pobanz

In sip.conf (assuming you are using SIP) you can set call-limit=10. This 
would limit the amount of calls the SIP account can place to 10. Another 
thing you can is have a macro compare the CID of the phone to where the call 
is going to. If they are the same dump the call and maybe have an agi that 
emails the sysadmin and or user of the issue. 



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Re: [asterisk-users] 64 bit asterisk

2007-10-18 Thread [EMAIL PROTECTED]
Nope it should just work. Just finished setting up 1.4 for the first
time in a while and just works. Been running 1.2 for the longest time
and same thing.

On 10/18/07, Wai Wu [EMAIL PROTECTED] wrote:
 Hi list,

 I just installed 64 bit Linux, and ready to install Asterisk through
 source on it. Are there any settings have to change to build 64 bit
 Asterisk? Thnx a million.


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Re: [asterisk-users] IAX2: Calls answered before extension is tested?

2007-10-18 Thread [EMAIL PROTECTED]
So your problem is:

-- IAX2/alanb-3 answered SIP/101-081d1050

Except the remote end didn't actually answer the call? The problem is
your remote end... its answering the call. All the IAX hardphones I've
seen don't seem to be the highest of quality honestly.

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