Re: [asterisk-users] [Zaptel] Why no port to Windos?

2007-12-15 Thread Time Bandit
 Which has existed, in one form or another, for years. I was using a
 voice enabled faxmodem a decade ago to answer my phone. The software
 that came with it (don't remember the name, but WinFax also does/did
 this) even allowed for a simple IVR, for mailbox selection and whatnot.
 The only things it didn't do that asterisk does (and would be useful to
 the average Joe) was support multiple phones/extensions and send
 voicemail messages via email.

I think what you are looking for is named SuperVoice :
http://www.supervoice.com/asp/products_supervoice_fax_products.asp

I was using it to receive faxes and voicemail. It didn't email me my
fax and/or voicemail but it would page me the number of faxes and
voicemail I had everytime it received one or the other.

The only problem I was having was that, since running on Win9x,
sometime my phone line would stay busy and that would signal the time
to go home and reboot the computer.

Maybe some people would like Asterisk for windows, but I would not
touch it with a ten foot pole :)

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Re: [asterisk-users] Cisco Directory Format

2007-09-01 Thread Time Bandit
 A little off topic (sorry..:) ) but anyone know what format Cisco phones
 use for their contact dirctories. I want to set up my contact lists on
 the phone, and cannot seem to get any info on it. I am working with a
 7970 on Asterisk 1.4.8.
7940 and 7960 use this format of XML file (probably the same on 7970)

CiscoIPPhoneDirectory
  TitleEmployee directory/Title
  PromptOpen Source Rock/Prompt
  DirectoryEntry
NameEmployee A/Name
Telephone7001/Telephone
  /DirectoryEntry
  DirectoryEntry
NameEmployee B/Name
Telephone7002/Telephone
  /DirectoryEntry
/CiscoIPPhoneDirectory

Check also Open 79XX XML Directory : http://web.csma.biz/apps/xml_xmldir.php

hope that help

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Re: [asterisk-users] Free sitting

2007-08-06 Thread Time Bandit
 In fact, my questions are more about usage than about technical background.
 For instance, I doubt a user will log his system off when leaving : some
 don't even turn their PC off.


 Does anyone has an experience to share about that ?
When I tried it, when a user login at a phone, it replaced any
previously logged one.

hope that help

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Re: [asterisk-users] What is the best softphone work with Asterisk

2007-07-24 Thread Time Bandit
 I need to configure a softphone to be client and use
 it with Asterisk, which is the recommended one? Is it
 iax2?

You can try my IAX2 softphone for windows :
http://www.marccharbonneau.com/asterisk/mediaxphone.php

Hope it fits your need

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Re: [asterisk-users] Best and easiest soft phone for my Dad..

2007-07-22 Thread Time Bandit
 Can anyone post a sample of whats needed in iax.conf for an IAX UA to be
 able to make and receive calls?

[7011]
type=friend
secret=S0m3S3cur3P4ssw0rd
qualify=no
notransfer=yes
[EMAIL PROTECTED]
host=dynamic
disallow=all
allow=ulaw,alaw,gsm
context=from-internal
callerid=Marc Charbonneau 7011

hth

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Re: [asterisk-users] Best and easiest soft phone for my Dad..

2007-07-21 Thread Time Bandit
 So I am looking for a softphone thats really simple to setup and as
 foolproof as possible..

 If SIP is likely to be problematic to setup then I have no problem
 getting him to use IAX but will need suggestions of which IAX softphone
 to use and also how to configure it in the iax.conf (haven't done this
 before)..
You don't specify if he's on Windows, Linux or OSX. But if he is on
Windows, you can try my softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php

There is a version using INI file, so you can put all the settings
then zip it and send it to him already configured.

hth

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Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?

2007-05-15 Thread Time Bandit

Could I rewrite this in Delphi instead?


I never used Delphi to write an AGI but I've seen a class in
FreePascal that you could probably use as a base :
http://www.automated.it/asterisk/fpc-agi.html

hth
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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

2007-05-04 Thread Time Bandit

A. Yes, I have the cojones. He never mentioned what platform it was for.
We need something like this for Linux. I got all excited about it only
to be terribly disappointed when I unpacked it.



From the original announcement : It runs on any modern flavor of Windows.


It is not like if he said runs on windows or better, then Linux
would seem appropriate ;)

You have at least two option beside having a windows machine :
1 - try it in WINE
2 - install Windows in a vmware machine (only way to run windows
without ever rebooting your machine)

hth
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Re: [asterisk-users] SugarCRM, NO!, Foxpro, SI?

2007-04-30 Thread Time Bandit

 First - vtiger is available for those who don't like the SugarCRM
 licensing.

It's not a licensing complaint.  At least that has not surfaced.  It is more 
that the
programmer does not seem to be comfortable with SugarCRM, MySQL and php.
Biggest compliant about sugar is - hard to configure, does not work with latest
php.

Have a look at vTiger then (fork of SugarCRM). Works with latest PHP
and MySQL, easy to configure and is free : http://www.vtiger.com/


hth
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Re: [asterisk-users] call dispatching - legacy application

2007-04-26 Thread Time Bandit

need to preprocess
1) incoming call get caller id lookup some info in my db,
2) based on the result dispatch the call to the right operator

step 1 is ok I developped a small .php script that connect manager and
parse events, now I have to tell AAH do dispatch call to the right
operator
From your incoming context, call an AGI and pass it the CallerID. In

the AGI, query your DB to find your destination then set an Asterisk
variable with the destination. In your dialplan, take that destination
and dial it. Something like

exten = s,1,Answer   ; Answer the line
exten = s,n,Wait,1
exten = s,n,AGI(aginame.php,${CALLERID(num)})
exten = s,n,Dial(Local/${MYDESTINATION})
exten = s,n,Hangup()

N.B.: Code not tested and written from the top of my head

N.B.2: Since you're using AAH, you should put that in extension_custom.conf

Also, have a look at the PHPAGI class : http://phpagi.sourceforge.net/

hth
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Re: [asterisk-users] Changing Voice from Male to Female

2007-04-26 Thread Time Bandit

Hi List,
I wanted to know if anyone knew of a way with asterisk to switch the voice
of a caller from male to female or vice versa.

http://www.lobstertech.com/code/voicechanger/

hth
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Re: [asterisk-users] Transfer via CTI

2007-04-20 Thread Time Bandit

Any ideas on this?

Closest thing that comes to mind is FOP : http://www.asternic.org/

hth
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Re: [asterisk-users] Remastering asterisk

2007-04-07 Thread Time Bandit

Anyone have an idea to re master centos,in other worlds I have an asterisk
on  centos with all libraries and modules,how can I make it as an iso image
?

Have a look at Kickstart

hth
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Re: [asterisk-users] Doorphone

2007-03-28 Thread Time Bandit

Responsibility for answering the door is shared by the entire office.  But A) noone wants 
their phone to ring, there's a door chime) and B) noone specific will accept 
responsibility for answering the door.  So, we need a solution that follow I'm 
answering the door now, these are the buttons I push.


So, when someone is at the door, you call whatever extension to get to
the door intercom, talk to them, then you decide to open it. You
hangup, then dial an extension that does only this, unlock the door.
Something like

[door-opener]
exten = 555,1,System(script_to_unlock_door.sh)
exten = 555,n,Hangup()

If you really don't want to have to dial a second extension, look at
applicationmap in features.conf
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf

hth
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Re: [asterisk-users] Doorphone

2007-03-27 Thread Time Bandit

On 3/27/07, Ray Wadkins [EMAIL PROTECTED] wrote:

I looked at a call queue, but it didn't seem to work the way I want.  Agents 
need to log into the queue to get calls, seemingly.  Of course, I only stopped 
on the topic for a short period.  with the meetme conference, anyone can answer 
the door from any phone by dialing the conference extension, just not open the 
door.



You can have static agents so they don't have to login, check
http://www.voip-info.org/wiki-Asterisk+call+queues

Wondering why you don't just dial multiple-phones, like this
Dial(SIP/7001SIP/7002SIP/7003)

The first one that answer the call is the lucky one. That way, your
DTMF signals would work.

hth
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Re: [asterisk-users] Voicemail mailbox number passed in connection?

2007-03-21 Thread Time Bandit

Does anyone know how to configure a SIP phone to pass the mailbox number to
the voicemail service when dialing?  I would like to press the message
waiting lamp and be prompted for my password instead of mailbox number.
Can this be passed in the set-up call or based on caller-id?


based on callerID with something like this :

exten = *97,1,Answer
exten = *97,n,Wait(1)
exten = *97,n,VoicemailMain([EMAIL PROTECTED])
exten = *97,n,Hangup()

then just configure your phone to point to *97 (or whatever you choose
as this extension)

hth
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Re: [asterisk-users] play file and action only stop if one defined key has been pressed

2007-03-09 Thread Time Bandit

I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and then
the file will be played from start again.

I would like that the play of file is only stopped if the user has pressed the
key 3.

What for an command can i use to make this happened?


check http://www.voip-info.org/wiki-Asterisk+cmd+Background

I think the m option is what you are looking for

hth
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Re: [asterisk-users] Summary of Trixbox vs. custom install

2007-02-19 Thread Time Bandit

You mean compiling raw tar.gz or SRPMS? And where do you download
them from? Trixbox site or the original vendors' sites?

I just download the tarball from asterisk.org and compile it. Trixbox
is not a special version of Asterisk, it is just an easy way to
install Asterisk, FreePBX, FOP and a bunch of other packages.

hth
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Re: [asterisk-users] Native format prompts

2007-02-18 Thread Time Bandit

I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul  .pcm . Which should I use so
Asterisk recognises them as native uLaw files
From what I know, .ulaw


hth
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Re: [asterisk-users] Summary of Trixbox vs. custom install

2007-02-18 Thread Time Bandit

I also include a consideration from mine: I would happily use
Trixbox, because I did FreePBX setup once and it was a real pain, but
I'm very frightened by a few issues:

1) Trixbox Macho installation that installs everything without
asking. I, for example, would like to use software RAID (maybe it's
wrong with Asterisk, but I want to do it!). I wouldn't like doing it
manually after Trixbox installation. I would like to have an
installer doing it for me. Centos (ex redhat) installer does it, so
why Trixbox choose to install everything without prompting?

You can just install CentOS with RAID and whatever you want, then use
the Trixbox tar package instead of the ISO. Still, why on earth did
the Trixbox team didn't leave the option of doing a custom install
with the ISO ?


2) How easy it is to find Trixbox SRPMS?  Is it possible to compile
new software (i.e. Asterisk or FreePBX, etc..) on Trixbox without
having to rewrite all the configuration files, changing all paths,
all permissions, and so on...

You can update Asterisk/zaptel/whatever by just downloading the source
and compiling it. My home system was installed with [EMAIL PROTECTED] on version
0.8 (if memory serves me right) and I upgraded Asterisk and Zaptel to
version 1.0.10 by downloading and compiling. I know, this is a really
old version and I should upgrade, but hey, it is doing everything I
need and it is stable (uptime of 315 days).

IMHO, Trixbox can me customized alot, but you need to know where and
what to modify. I believe that if you know enough about how Asterisk
work, you can get around Trixbox limitations. One thing to remember is
that the files you can modify are the _custom.conf files. Never touch
the _additional.conf files, they will get overwritten next time you
click Apply changes in the GUI. The normal base files (sip.conf.
iax.conf, etc) can be modified since the GUI doesn't touch them.

But I also think that there is nothing that can beat a plain install
as far as customization go. YMMV
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Re: [asterisk-users] Asterisk dual contexts stupidity

2007-01-31 Thread Time Bandit

Significant albeit insanely stupid Asstricks message:

2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)

Any thoughts

What Error ?  it says DEBUG

This just tell you that the user pressed '2'

Actually, the first time I read that message I was laughing :)

Only in opensource product you have the priviledge of having funny message

hth
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Re: [asterisk-users] Testing IVR / Callcenter applications

2007-01-31 Thread Time Bandit

We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??

Just use an IAX or SIP thrunk to/from another Asterisk.

there is no real difference from Asterisk's stand point if the call
comes from IAX, SIP or ZAP

hth
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Re: [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

2007-01-26 Thread Time Bandit

How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.

the keyword is context

boss extension : 4321
secretary exten : 4322
in sip.conf for the secretary config, put her phone in the context
secretary-context
for other callers (PSTN lines, other office exten, etc) put them in
context normal-people-context

[normal-people-context]
exten = 4321,1,Dial(SIP/4322)

[secretary-context]
exten = 4321,1,Dial(SIP/4321)

like this, when someone dials 4321, they will reach his secretary,
except when the secretary dials it, she will reach him.

This is just an example written from the top of my head on a friday
afternoon so it is not tested, etc  :)

hth
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Re: [asterisk-users] dialplan and *

2007-01-25 Thread Time Bandit

exten = ,n,Queue(|t|||300)
exten = *,1,Macro(agent-add,,)
exten = **,1,Macro(agent-del,,)

So my question is , what means these one/two asteriks (*,**
).Maybe it is like priority.?

It means that to login as an agent on the queue you have to dial
* and to logout you dial **.

hth
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Re: [asterisk-users] Wanpipe 2.3.4-2 + kernel 2.6.19 = problems

2007-01-15 Thread Time Bandit

This is the error i got. I've grepped through all of my include/linux/
wanpipe_includes.h files i have on my server (there is actually a
couple of them), and replaced config.h with autoconf.h, but still i
get the same error. Looks like I'm unable to locate the include/linux/
wanpipe_includes.h file wanpipe is actually looking for. Is there a
patch or a newer version of wanpipe that has this issue solved?



From the changelog of 2.3.4-4 released on 2007-01-09

(ftp://ftp.sangoma.com/linux/current_wanpipe/ChangeLog.stable)

- Updates for 2.6.18 and 2.6.19 kernels.

hth
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Re: [asterisk-users] phpagi transfer example

2007-01-15 Thread Time Bandit

Ok, how can i do the transfer from the caller to $keys ?

Probably by using a goto :
http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#goto

hth
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Re: FW: [asterisk-users] Get dialed numbers in AGI

2007-01-12 Thread Time Bandit

All the variables here was my_var, it worked for GET VARIABLE but didn't for
SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert
my_var value into a db?

- What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for
digits.
- STDIN shoudn't get the result of READ or GET VARIABLE? Where these values
go?

For AGI in PHP I always use this : http://phpagi.sourceforge.net/

Would make your life easier

hth
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Re: [asterisk-users] getting tones during conversation

2007-01-10 Thread Time Bandit

after the Dial has connected, I want the caller (on a SIP phone) to be
able to press keys in order to record call status.  is this possible?


Have a look here :
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf

applicationmap is what you are looking for

hth
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Re: [asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

2007-01-10 Thread Time Bandit

The phone in question just prepended 010whatever to ALL phone numbers
dialled, which makes it pretty crappy to use with a line that does not
allow for network selection codes, or on lines that need a 0 for a
POTS line.

You could use it as an extension on Asterisk and strip that pre-pended
number from your dialed number.

Nice way to screw their attempt to screw you :)
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Re: [asterisk-users] Sangoma Remora A202

2007-01-03 Thread Time Bandit

Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my Dell GX260,
but the power light on the front of the box just blinks and won't
power up.

Maybe your card is not properly seated.


seems to have a lack of documentation, but it may just be me

It is just you ;)

http://wiki.sangoma.com/

If you still have problems with the card, contact Sangoma, they have
very good customer support : http://www.sangoma.com/main/contact

hth
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Re: [asterisk-users] Sangoma Remora A202

2007-01-03 Thread Time Bandit

Thanks - that turned out to be the problem.  Well- one of those solutions.
I removed the blank and swapped the FXO module to the other port.  I don't
know if it was a bad port on the A200, but since I don't plan on using it, I
won't worry about it- just regret it in a year when I get a second FXO
module ;)


No you won't, since Sangoma cards come with a 5 year warranty ;)

Glad you fixed it
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Re: RE : [asterisk-users] Happy 2007!!!

2006-12-31 Thread Time Bandit

I wish you all a Happy 2007 filled with an almost-bug-free,
full-of-nice-features Asterisk 1.4 :c)
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Re: [asterisk-users] Background switch to different context

2006-12-28 Thread Time Bandit

I am using the Background() function to ask for the extension, but the
extensions are in a different context. Is there a way to tell Background()
to look for the entered extensions in another context other than the
currently running one?

in that context you can do
include = other-context

hth
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Re: [asterisk-users] Re: php agi trixbox help

2006-12-27 Thread Time Bandit

Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:

PHP Warning:  Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file:
No such file or directory in Unknown on line 0

yum install php-imap

hth
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Re: [asterisk-users] Insert 1+areacode for VOIP calls

2006-12-21 Thread Time Bandit

Is there a way I can create a _NXX extension and insert 1 and areacode
when dialing?

exten = _NXX,1,Set(CALLERID(num)=6162997590)
exten = _NXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1514${EXTEN})

replace 514 with your area code

hth
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Re: [asterisk-users] AGI Help Please

2006-12-20 Thread Time Bandit

Below are a few errors in the script and on a google search, although I
found people with the same error, I didn't find a resolution.

Any thoughts on what is causing this error?
Any thoughts as to why the output is not showing on the CLI without doing a
debug?

snip


Content-type: text/html
X-Powered-By: PHP/4.3.9


These 2 lines should not be there.


AGI Tx  
AGI Rx  
AGI Tx  510 Invalid or unknown command
AGI Rx  
AGI Tx  510 Invalid or unknown command

These 2 errors are probably caused by the Content-type and X-Powered-By lines.


AGI Rx  VERBOSEThere have been
AGI Tx  510 Invalid or unknown command
AGI Rx  VERBOSE125 calls made
AGI Tx  510 Invalid or unknown command


According to this page http://www.voip-info.org/wiki/view/verbose
Usage: Verbose(message [level])

Also, you usually put error_reporting(0); at the top of the script
so you won't have warnings and errors confusing Asterisk.

I never wrote a PHP AGI without using this : http://phpagi.sourceforge.net/
so I can't help you much
You should give it a try, you might like it :)

hth
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Re: [asterisk-users] Need quality toll free 800 number over IAX?

2006-12-20 Thread Time Bandit

I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.



Any suggestions please?

Never used them but the rates seems ok : http://www.les.net/
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Re: [asterisk-users] IBM Server / USB Ports

2006-12-19 Thread Time Bandit

So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!!

I would try moving the Digium card to another slot. Your Ethernet
controlled must be onboard and it share its IRQ with the slot where
the Digium board is.

hth
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Re: [asterisk-users] Need help getting started with asterisk

2006-12-19 Thread Time Bandit

Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all.  Is there some explicit thing I
need to put in to get the caller ID?

callerid=asreceived
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Re: [asterisk-users] Need help getting started with asterisk

2006-12-18 Thread Time Bandit

camille*CLI zap show channels
   Chan Extension  Context Language   MusicOnHold
 pseudoincomingen


This should show something like this :

panoramix*CLI zap show channels
  Chan Extension  Context Language   MusicOnHold
pseudofrom-pstn   en
 1from-pstn   en

so something is missing as Asterisk doesn't see your Zap channel

what does your zapata.conf looks like ?
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Re: [asterisk-users] AGI and php simple example

2006-12-17 Thread Time Bandit

I've read http://www.voip-info.org/wiki-Asterisk+AGI+php but i can't
understand how to play sounds and read DTMF digits...


Have a look at this : http://phpagi.sourceforge.net/

hth
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Re: [asterisk-users] Dial 9 For Outside Line?

2006-12-17 Thread Time Bandit

exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})


Just add a 9 in front, like this :

exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

hth
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Re: [asterisk-users] Dial 9 For Outside Line?

2006-12-17 Thread Time Bandit

Just add a 9 in front, like this :

exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})


Oups, pressed Send too fast, here is take 2

exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:2})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
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Re: [asterisk-users] Linux distro + Asterisk or Trixbox?

2006-12-16 Thread Time Bandit

I've been doing a lot of playing, and a lot of reading, and it seems people
are split as to whereas if they're running their favorite Linux distro and
asterisk or Trixbox.  I'm getting closer to really looking at a production
environment and I'm just looking for any opinions.  I'm really enjoying
learning linux and asterisk, so initial ease of use isn't really a huge
benefit to me.  In the end stability and upgradeability will be my main
concerns.

My favorite for stability and upgradeability is CentOS + Asterisk plain install

As a proof, here is what I get on my home PBX
[EMAIL PROTECTED] root]# uptime
10:04:01  up 250 days, 19:53,  1 user,  load average: 0.00, 0.00, 0.00

I think that speaks for itself :c)

I'm not saying Trixbox is not stable (since it is based on CentOS),
but it is not as customizable as a plain install (IMHO)

YMMV
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Re: [asterisk-users] TDM2400

2006-12-11 Thread Time Bandit

 [channels]
 context=default
 signalling=fxs_ls
 ;channel=1-16
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 restrictcid=no
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 group=1
 ;accountcode=lss0101
 answeronpolarityswitch=yes
 hanguponpolarityswitch=yes

To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.

hth
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Re: [asterisk-users] PCI, PCI-X and PCI-e -- Server / Interface Card Selection

2006-12-09 Thread Time Bandit

  I can't risk spending a few thousand just to reach the
  conclusion that Digium's PRI or BRI cards do not work
  with a particular system's PCI-X slots/bus... Or, worse,
  staying with a dead card / system board in my hands ! :-(

  Anyone ?

I don't know about Digium cards, but I just installed a Sangoma A101
card into an IBM server in a PCI-X slot and it is working perfectly.

You should ask Digium

hth
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Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

2006-12-08 Thread Time Bandit

Does there seem to be a popular Linux distro folks use specifically for
Asterisk?  I'd like to move off of FreeBSD but I'm not too familiar with
Linux distros.  In particular, I'm looking for a free, stable, well
supported distro that has a friendly community.  Any advice appreciated.

CentOS works well for me : http://www.centos.org/
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Re: [asterisk-users] any possibility of Vonage Integration

2006-12-06 Thread Time Bandit

$25/month will buy you close to 50 hours of urban SIP termination,
it's down to half a cent in some of the big cities.   Are you
going to average 50 hours on the phone each month?   Some people
do, but most don't.   (Otherwise Vonage could not even pretend it is
going to make money.)

You don't have a teenager in your home I guess ;)

The teenager girl in my home can easily make more than 3000 minutes of
call in a month !
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Re: [asterisk-users] How to stop Asterisk to pick up incoming PSTN signal

2006-12-05 Thread Time Bandit

Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the
functionality to make the call out?

[from-pstn]
exten = s,1,Hangup
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Re: [asterisk-users] How can i processed with Call Snooping,

2006-12-04 Thread Time Bandit

How can i Processed the call Snooping, it my fifth Requesting and posting
to Users, Nobody  replies it,,,

see http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy

hth
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Re: [asterisk-users] extension launch into AGI

2006-11-30 Thread Time Bandit

I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when picking up the
hand set, the AGI is launched without dialing any digits.

check http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
keyword is : immediate
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Re: [asterisk-users] Live call monitoring

2006-11-30 Thread Time Bandit

What I'd like to implement, ideally, is that once an incoming call is
transferred to a particular operator, the system also calls a manager
who can monitor silently.

I think you are looking for this :
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
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Re: [asterisk-users] Loosing IAX connection between offices

2006-11-30 Thread Time Bandit

if its a version 5 or higher, that wont be an option, but if its not,
give openwrt or ddwrt a try.

Actually, this is no longer true (at least for WRT54G), see
http://en.wikipedia.org/wiki/DD-WRT for the official list of supported
models
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Re: [asterisk-users] 2nd attempt - Return code - How to?

2006-11-30 Thread Time Bandit

Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not
registered

from http://voip-info.org/wiki/view/Asterisk+functions :
Functions in the below list are marked in red if they are only
available in version 1.4 and higher.

And STAT is marked in red so I guess you're not running 1.4
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Re: [asterisk-users] AGI PHP Issues (Not new to Asterisk but new to AGI)

2006-11-29 Thread Time Bandit

I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC).  I have dabbled with PHP but I am by no means
a professional standard developer.

Can't really say what is wrong with your code since I never did an AGI
in PHP without this class : http://phpagi.sourceforge.net/

This should make it more easy

hth
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Re: [asterisk-users] Re: Rewriting caller ID from database?

2006-11-25 Thread Time Bandit

I use some custom scripts to do database lookups and rewrite CallerID
information.  Everything works fine with regard to the CID name, however
my Cisco 7960 and Linksys SPA-942 phones do not display the calling
number. Instead, they display the called number.  This makes the phone's
call return feature not work. The calling number and name are both
properly displayed on all of the softphone clients that I've tried.

Here's the format I'm using to set the CallerID.

SET CALLERID JONES DARYL A6508701826

If you're using Asterisk 1.2, see this page :
http://www.voip-info.org/wiki/view/Setting+Callerid

hth
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Re: [asterisk-users] Asterisk and TDM400P ?

2006-11-24 Thread Time Bandit

for see if the card answer, what is the process ?

since your port is configured to be in the interne context, just add
this to this context

exten = s,1,Answer
exten = s,2,Playback(tt-monkeys)
exten = s,3,Hangup

watch the console and dial-in. if you get monkeys screaming at you, it worked !

hth
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Re: [asterisk-users] Asterisk incoming call behaviour

2006-11-23 Thread Time Bandit

I am using asterisk to receive call from a DID provider . In configured
everything in freepbx properly and its working . I forwarded incoming calls
from did to a certain extension . Now  i tried calling from another sip
provider to this box , when i call from other provider to my DID number
then call reaches asterisk and is sent to configured extension ..  however
if  the extension hangs up without picking then also i am being billed at
sip provider ( outgoing one ) . In simple words when people call me then
they ( other people ) are billed even if configured extension isnt picked up
and hangs the phone. Normally when you call a person and
they hang up then you arent charged . Is
this asterisk behaviour or is it freepbx dialplan
the culprit here ?

check your the context into which the calls are coming. if you have an
answer line, there is the culprit

hth
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Re: [asterisk-users] Rewriting caller ID from database?

2006-11-22 Thread Time Bandit

Most of our customers have generic names like Hospital, so I need to
rewrite their caller ID name by looking up the number in a database on the
Asterisk server, and rewriting the name such as Reading Hospital so that
we know who's calling.

Any idea if this can be done with Asterisk, and how to do it?


I made a simple PHP AGI that takes the phone number and query a MySQL
table to find the name assigned to this number.

I still need to make a web interface to enter/modify the list but
phpMyAdmin do the job for now.

If you want it, just let me know.
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Re: [asterisk-users] Asterisk as an analog call recording solution (was: Recording outbound analog calls with X100P)

2006-11-16 Thread Time Bandit

Thank you for the confirmation and the warning about disk space.

Now I need to decide between the Sangoma A20202 and the Digium TDM2411.
I'm leaning heavily toward the Sangoma card for the following reasons:

- It doesn't require a 12V power connector for the operation of FXS modules.

Maybe, never used it so I can't confirm, but I doubt it since I'm
wondering where it would pull the needed power for the FXS (from the
PCI bus ???)


- It is compatible with 5v and 3.3v PCI buses.
- It shares PCI interrupts properly.
- It maintains a single synchronous PCI interface for all FXO/FXS ports
(additional daughterboards are added to a backplane bus connector)
- It has a better form factor.

Depends, if you want to put more than 4 ports in a 1U server, should
be easier to put a full-length TDM2400P than try to squeeze 2 A200
cards side-by-side.


- It is less expensive.

These are rather compelling reasons, but I'd still like some feedback
from the list prior to making a purchase.

I've seen lots of good comments on the Sangoma cards and on their
support. I never used the A200 but I can confirm that Sangoma say they
support FAX calls, while Digium say that the TDM2400P was only
designed for voice. YMMV


Both cards offer hardware echo cancellation for an additional cost.  Do
the benefits of hardware EC justify the expense in this scenario?

Usually when you bridge 2 ZAP channels you don't need echo cancel. I
would buy without the echo canceller and if you really have echo that
you can't get rid of with software echo can, you could add the
hardware echo can after. With Digium's TDM2400P you can add the echo
can after. With Sangoma, I think you have to exchange the card for one
with it. You should call Sangoma to confirm
(http://www.sangoma.com/main/contact)


Thank you,

You're welcome
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Re: [asterisk-users] Recording outbound analog calls with X100P

2006-11-15 Thread Time Bandit

Is it possible to record outbound analog calls using an X100P?

I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users.  I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
A200s with 4 FXO and 4 FXS modules.  The FXO modules would be connected
to the existing PBX and the FXS modules to the existing analog phones.
Then with a simple dialplan, all inbound and outbound calls could be
recorded by Monitor.

I wanted to mock this up using some X100Ps that I had laying around, but
found that I could only record inbound calls.  I believe that I need an
FXS interface to record outbound analog calls but my past experience is
with T1 interfaces, so I could be mistaken.


Of course you can, if you have 4 FXO and 4 FXS, you could make a
really simple dialplan and record the calls that pass through it,
incoming or outgoing, and the users wouldn't even know that there is a
pbx between them and the PSTN.

You will need a lot of space to keep them all, but you could make a
simple cron job that would erase any recording older then, say, 2
months.

Also, you would have the benefit of having CDR

hth
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Re: [asterisk-users] Problem with internet down

2006-11-13 Thread Time Bandit

I believe that the problem really is fault of DNS lookups, but as I
should proceed for resolve that??

see the first point at
http://www.voip-info.org/wiki/view/Asterisk+administration

The best solution for now is probably to have a caching dns server on
your Asterisk box or in your LAN
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Re: [asterisk-users] Ringing phones

2006-11-08 Thread Time Bandit

Apologies.. we are using a sangom 4 port FXO card.   It used to work
(or so the company claims that has the PBX), but they are saying it
stopped.. yet nothing has changed on the PBX system.  I have verified
it IS picking up and then passing the call onto the ringgroup (hence
taking it out of the phone companies domain).

Matt,

check in your incoming context that you don't have an Answer before
you dial the ringgroup.

If you don't answer and just dial the ringgroup, Asterisk won't pickup
the incoming call until a phone in the ringgroup answers it.

hth
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Re: [asterisk-users] Follow Me problems

2006-11-07 Thread Time Bandit

Today we appear to have discovered our first bug.  We have an extension
setup to followme by ringing that extension + an external cell #
(ringall).  If nobody answers after 20 seconds the destination if no
answer is set to go to the extensions voicemail in the followme module.
The problem is it just keeps ringing forever.  If we delete the followme it
forwards to the voicemail as per the default SIP extension configuration
with voicemail enabled.

Anyone run into this?  Is there a workaround?  Any advice would be greatly
appreciated as always.

Our configuration is:
Supermicro Pentium D 2.66 Server with 2x512MB Memory
3ware 8006-2LP Hardware RAID 1
Sangoma A200D with 8fxo (latest firmware/drivers as of last week)
CentOS 4.4
Asterisk 1.2.13
Zaptel 1.2.10
FreePBX 2.1.3


When Asterisk dial the Cell phone, it goes out on the ZAP channel
(Sangoma A200D), so as soon as it hit that channel, the call is
considered answered even if the cell phone never actually pickup the
call. I didn't play with the followme module myself but that is what
I suspect is happening. Just watch the console and you should see
something like Zap/1-1 answered ...

hth
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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

2006-10-31 Thread Time Bandit

After installing properly when opening in the webpage it is not parsing the
index.php for the AMP. My Database is MySQL.and web server is Apache 2.2.


Please let me know is this configuration problem or this is the problem with
Apache (Apache 2.2) .

The problem is probably that you didn't install PHP

yum install php
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Re: [asterisk-users] Compatability

2006-10-31 Thread Time Bandit

I have a new client who has an existing  Asterisk  PABX and is looking
for us to install a TE110P for him, However he has a Dell SC420 and I
have never used one before.
I have had no problems with any other Dell servers which we use almost
exclusively.

Has anyone had any good/bad experiences with the SC420 in relation with
Digium cards?

According to Digium this model is partially incompatible :
http://www.digium.com/en/docs/misc/compatibility_notes.php

I would suggest a Sangoma card if you want to avoid problem with that server
http://www.sangoma.com/datasheets/p_aft-et1-specs

hth
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Re: [asterisk-users] No ring tone when using IAX

2006-10-31 Thread Time Bandit

Then what would be a better solution?

Usually the IAX phone will play you a ring tone until the other end
answer. If you're phone doesn't do it, then it is a flaw in that
phone.

What phone is this ?
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Re: [asterisk-users] operator console

2006-10-30 Thread Time Bandit

...but I'll need to give the users a good mean to see
what's going on,
who is busy,
easy transfer with click here and there,
easy conference,
easy queue handler,
easy way to see/use many lines at the same time

is there any best console they can use?

Have a look at FOP : http://www.asternic.org/
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Re: [asterisk-users] Wildcard X100P Suport

2006-10-30 Thread Time Bandit

Is the Wildcard X100P still supported?  I have one sitting around that I
bought 3+ years ago and never used it.  I need the functionality now.
Before I run off and buy something new, I'm curious if this will just
work.

It still works with the latest Zaptel (1.2.10)


I also have an old TDM400P with 2 FXS modules that I bought at the same
time.  Then, there was no FXO module for the 400P.  Will a TDM400P this
old support a new X100M?

I don't think that something changed, so it should work. You should
contact Digium to be 100% shure

hth
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Re: [asterisk-users] voicemail usernames can't begin with j letter?

2006-10-23 Thread Time Bandit

Thanks to all that replayed, I made like Mr Watkins told me, and my problem is
apparently solved, although, because of the usage of the syntax
VoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and
vm-isunavail, while before were only played vm-intro and beep.
Is there a way to disable this two other files that get played every time?

see http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
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Re: [asterisk-users] Getting started with sample dial plans

2006-10-20 Thread Time Bandit

Now I'm ready to begin playing with dial plans and am having a difficult
time getting started.

You may want to read the book :
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

That should help you
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Re: [asterisk-users] Why is this happening?

2006-10-17 Thread Time Bandit

You've got a very poor grasp on how things work.  Please don't pretend to know
what you're talking about.

# netstat -apn | grep :80
tcp0  0 0.0.0.0:80  0.0.0.0:*   LISTEN
782/httpd
tcp0  0 204.xxx.yyy.188:8080.xxx.yyy.167:58620
ESTABLISHED 814/httpd
tcp0  0 204.xxx.yyy.188:8062.xxx.yyy.15:55384
ESTABLISHED 1068/httpd
tcp0  0 204.xxx.yyy.188:80165.xxx.yyy.230:4392
ESTABLISHED 1084/httpd
tcp0  0 204.xxx.yyy.188:8065.xxx.yyy.111:6982
TIME_WAIT   -
tcp0  0 204.xxx.yyy.188:80200.xxx.yyy.43:8198
ESTABLISHED 817/httpd
tcp0  0 204.xxx.yyy.188:80165.xxx.yyy.230:4304
ESTABLISHED 815/httpd

As you can see, I am *still* listening on port 80 and have numerous
connections from different systems, even numerous connections from the same
system.

I am really sorry, I've read that explanation somewhere and it made
sense. Now that I've been corrected, I won't make that same mistake
again.

Please excuse me.

The one that never did a mistake, never did anything
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Re: [asterisk-users] Remote UNIX connection, Remote UNIX disconnected displayed every second

2006-10-16 Thread Time Bandit

Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?

You probably have some script that use the console to query something,
like the WebMeetme application.

hth
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Re: [asterisk-users] Why is this happening?

2006-10-16 Thread Time Bandit

Why is it running on port 1207?

because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at a time.

Pretty basic networking stuff I think :c)
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Re: [asterisk-users] Why is this happening?

2006-10-16 Thread Time Bandit

Thanks for the answer, but I don't buy it.  There are currently 0
calls up on that bridge, while another connection which has calls up
on it is on Port 4569.. please try again.  IAX2 is suppose to run on
ONLY one port.. this is why it is so nice for use in firewall
situations.


It doesn't change a thing !

Same thing happens with a webserver. It listen for connections on port
80 (default port) and when a connection comes in, it is handed to
another free port on the server so the main server can continue
listening on port 80. Same thing with FTP, etc. All TCP servers that
accept more than one connection

I think that what iax2 show peers display is the remote port from
which the client connected. iaxclient library defaults to using port
4569 as the originating port but there is a function to specify
another port.

Check on your machine while you're surfing the web, your browser
doesn't use port 80 as the originating port. Connect to an FTP server
and check your netstats, you'll see that you're not connected to port
21 on the remote server
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Re: Re[2]: [asterisk-users] Why is this happening?

2006-10-16 Thread Time Bandit

On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:

OMG, please read more about network ports.

Could you tell me what is wrong with my explanation ?
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Re: [asterisk-users] OT: Hand free solution recommandation

2006-10-10 Thread Time Bandit

Ideal would be a headset audio+microphone with RJ11 4p female that we
could plug into the handset cable of any IP phone, or a converter
2xjack2,5mm female  RJ11 4p female -which seems not to exist-.

What are you recommanding/using/installing in such case?

I don't know if it would work on any phone, but it works on Cisco
7940/7960 : http://www.mml.uni-hannover.de/einhorn/headset/index_e.html

hth
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Re: [asterisk-users] MODEM (data) througt asterisk ?

2006-10-05 Thread Time Bandit

Is it possible to connect a modem to a remote service through asterisk ?
Basicly to ilustrate : Accounting department need to connect with analog
modem to their bank to order some wire transfert.

Modem - Chanel Bank FXS - Asterisk - TDM2400 FXO - Modem in
remote site.


If you get it working you're lucky. Digium's official statement on the
TDM2400 is that card as been designed for voice calls, we don't
support data calls

You would have better luck with a Sangoma A200

hth
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Re: [asterisk-users] pop a web page with DID in url

2006-10-05 Thread Time Bandit

I'm looking to do this.
When a call comes in to an agent in a queue, pop a web page like this
http://www.mydomain.com/cgi-bin/script.cgi?did=952900
Where did is the number the caller dialed to reach the system in the
first place.

I know Hudlite can do this we caller ID, but the DID feature is not
there yet.

Does anyone have any other software they know of that can do this?


Some softphones support handling URL when you pickup the call. You can
set that URL to anything you want from the dialplan. shameless-plug
My MediaX softphone (current beta version) support it. Let me know if
you want to try it /shameless-plug

hth
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Re: [asterisk-users] Call Interception

2006-10-04 Thread Time Bandit

I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.

have a look at these :

http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge

and

http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy

hth
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Re: [asterisk-users] Dialplan Syslog

2006-10-04 Thread Time Bandit

It'd be cool if someone wrote a syslog() dialplan application for Asterisk 
*hint* *hint*


That could be usefull, but what is wrong with : System(logger Asterisk
can use syslog)  ?
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Re: [asterisk-users] Right way to prevent analog channel from answering the phone?

2006-09-27 Thread Time Bandit

Is there a more elegant way to tell it to answer/not answer on command?

Put your Zap line in a context that do just this :

s,1,Hangup()

hth
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Re: [asterisk-users] IAX phones?

2006-09-27 Thread Time Bandit

Just wondering if there are any IAX phones worthy of the name phone out
there -- looking for hard phones, but I suppose a Linux-based softphone
wouldn't, you know, hurt.  ;-)

Idefisk looks pretty nice and there is a Linux version :
http://www.asteriskguru.com/idefisk/

There is also iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/index.html

Also, check on iaxclient page : http://iaxclient.sourceforge.net/

hth
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Re: [asterisk-users] Variable that gives the SIP channel

2006-09-18 Thread Time Bandit

  What I would like to do is in my flash hook dialplan code to ass
something like Hangup(SIP/100-fe65), but where can I get that
SIP/100-fe65 ? Is there a variable set with this information available
in the dialplan ?


${CHANNEL}

have a look here : http://www.voip-info.org/wiki-Asterisk+variables

hth
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Re: [asterisk-users] Cisco 7940 Problem (Mess)

2006-09-18 Thread Time Bandit

A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll
work on my Asterisk box (outside of the FXO  FXS modules on the TDM card in
the Asterisk server, I only run SIP on the hardphones).

I don't know the phone's password (sound familiar?). - Have tried
everything, cisco, *##, etc  Nothing works.

You could factory-reset the phone.

try this
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml

and this
http://www.sokol-associates.com/?q=node/51

hth
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Re: [asterisk-users] Modem calls

2006-09-15 Thread Time Bandit

 I need to pass modem calls through a TDM400 card. Conecting the modem to
the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.

According to Digium, Fax calls (and modem calls) are not supported on
the TDM400 or TDM2400. They are designed for voice only. If you get it
to work, you're lucky.

Sangoma test all their cards with faxes, so maybe you should try their card.

For your problem, run zttest and adjust everything to try to obtain
100%, this may make it work.

hth
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Re: [asterisk-users] sound file length

2006-09-12 Thread Time Bandit

At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)

the record application doesn't seem to have any facilities to do that.

any ideas ?

use sox beep.wav -e stat and parse the output

man is your friend
google also :)
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Re: [asterisk-users] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration

2006-08-31 Thread Time Bandit

We have a problem in configuring Sangoma A104. We want the 2 ports to be
configured as E1 and the 2 ports as T1.


If I'm not mistaken, you can't do that with the A104D, that's why they
sold me 2 x A102 for the same price as a A104. Better check with
Sangoma.

hth
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Re: [asterisk-users] detecting a users number using the dialplan or AGI

2006-08-27 Thread Time Bandit

keeping track of the confno is easy since I created it,
but I don't know how to determine the user number of the last person that
joined the conference.

Is there a way to store this in a variable before they join the conference?
Or perhaps a way to detect the last user to join the conferences number?


Maybe by listing the users in the conference and parsing the output
something like : meetme list 87004

you will get an output like :
User #: 1  Channel: SIP/7004-1d3f (Admin)

hope this help
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Re: [asterisk-users] 7970 'LoadID incorrect' problem

2006-08-26 Thread Time Bandit

Does ANYONE have any clues?

Only played with 7940 and 7960, but I will try to help since nobody
comes forward


loadInformationSIP70.8-0-3S/loadInformation

Shouldn't that be something like P0S3-08-2-00 ?
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Re: [asterisk-users] Monitoring/Listening In

2006-08-24 Thread Time Bandit

I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?


http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
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Re: [asterisk-users] About IVR and Oracle

2006-08-23 Thread Time Bandit

On 8/23/06, Infobox Peru [EMAIL PROTECTED] wrote:

maybe you could make it with PHP and its driver for Oracle.


For PHP have a look here : http://phpagi.sourceforge.net/
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Re: [asterisk-users] Text to Speech

2006-08-21 Thread Time Bandit

N.B.: Please use plain text when sending to this list


Can someone recommend a good text to speech engine that is usable by Asterisk? 
I have tried the Festival one and it just doesn't cut it for commercial 
applications.



We are willing to pay for a good one that works. Anyone tried the ATT speech 
engine? The IBM ViaVoice sounds no better then Festival.


You have flite that is free and, IMHO better than festival
(http://nerdvittles.com/index.php?p=134).

I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting
for the Linux version as I don't have time to babysit a Windows server
:)

hth
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Re: [asterisk-users] Text to Speech

2006-08-21 Thread Time Bandit

All I can find for Flite is for AAH, does it work as well with plain
Asterisk? Is the setup the same?

Never tried it, but it should be the same.

Have a look here : http://dialogpalette.sourceforge.net/extras.html

hth
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Re: [asterisk-users] astbill white screen!!

2006-08-17 Thread Time Bandit

I've installed asterisk and astbill according with all recommendation
(mysql5, drupal included with astbill, php, apache2...).
When I write http://server_adress/astbill, I get a white screen page.
Browser doesn´t give me an error page, it just a white screen page.


you have to enable it in php settings.

Go in /etc/php.ini
- change setting error_reporting  to E_ALL
- change setting display_errors to On
- restart apache

now, at least, it will tell you what goes wrong

N.B.: display_errors should not be enabled on a production server

hth
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Re: [asterisk-users] Rookie question, trying to learn

2006-08-02 Thread Time Bandit

The problem a number of people are not entering the pin fast enough
,they are not given enough time to enter the PIN( I assume this is a
mailbox number)

looking at all the doc is seems everything is configurable, can some
one point me in the right direction of where to start looking?

check http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout

and http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ResponseTimeout

Hope this help
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Re: [asterisk-users] Examples of handeling input from phones with PHP

2006-07-18 Thread Time Bandit

Can anyone direct me to where I might find examples of handling
interactive input from a phone using PHP and AGI. I want to have someone
dial an extension and then have the system request input from the user,
take that input and put it into a database.


Start here : http://phpagi.sourceforge.net/
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Re: [asterisk-users] phpagi problem

2006-07-17 Thread Time Bandit

#!/usr/bin/php -q
 ?php
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-say_digits(62410);
$cid = $agi-get_variable(dir);
$agi-say_digits($cid);
 ?


I'm getting this error:

parse error, unexpected '=' on line 6


I don't know why you're getting this error, it parse correctly here.

But one thing is that the line $agi-say_digits($cid); won't work.
When you do a get_variable, the result you get is an array, and the
member holding the value is 'data'. So you have to write your line
like this : $agi-say_digits($cid['data']);

Check the documentation : http://phpagi.sourceforge.net/phpagi2/docs/

hth
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Re: [asterisk-users] Can one SIP extension be used for two phones?

2006-07-09 Thread Time Bandit

Picture this:

Exten = 100 #My Phone
Exten = 200 #MythPhone

Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another MythPhone to my other MythTV box upstairs.

Do i have to make a third extension and get the dialplan to call all
three extensions or can the second MythPhone instance also log into
Asterisk using the same extension?


Each phone needs its extension

But, you can have the dialplan ring more than one phone and the first
to pickup the call is the lucky one :)

like : Dial(SIP/100SIP/200SIP/300)

hth
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Re: [asterisk-users] Global variables and AGI

2006-07-09 Thread Time Bandit

Hi everyone,

I know that functions like set_variable and get_variable (using php with
phpagi) only apply to the channel variable. What I need to do is reset a
global variable I have in our system. I have a script that is going to
determine when this will happen, but I just have to make it happen.
Assuming that I cannot update the variable via the script, it is there a
way  I can make a call to the system, such as a call file, and place it
in the context of the dialplan that I need to change the variable? If
so, is there anything special I need in the call file for that to work?
Or is there a easier/better way to do this that I haven't thought of.

Reading this : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set
you need to use the g option

If you can't use it from phpagi, you could, at worst, set a local
variable from your AGI, then in the dialplan take the value of that
one and apply it to the global one using option g

hth
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