Re: [Asterisk-Users] AreskiCC

2005-05-16 Thread Stiffe
My 2 cents:
If I dont misunderstand it, I would guess youll have to read it
again and type in myroot AND mypassword  or what it says and NOT
your_login and your_password.
After that, you can change it to your preferred password etc...

Thats what I guess...But Im not sure.

Regards

//Stefan

On 5/16/05, Robson Ribeiro [EMAIL PROTECTED] wrote:
  
  
 
   
 
 Hi, 
 
   
 I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and
 Postgres 7.4. First of all the instructions are very confusing and hard to
 follow if you are not an expert. But, I managed to install it andobviously
 t doesn't work. The other instructions I found on wiki are a great effort
 but incomplete. Basically the first thing that happens is that when I load
 /areskicc/Public/index.php it refuses my username and passwork
 (AUTHENTICATION REFUSED, please check your login/password! ) which I guess
 is the same as the one I configured on defines.php right?) and after I
 reinsert it I get the error: Method Not Allowed. The requested method POST
 is not allowed for the URL /areskicc/Public/index2.php. 
   
 In any case, does anybody know of any better instructions on how to install
 and configure AreskiCC? 
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   
 http://lists.digium.com/mailman/listinfo/asterisk-users
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dial Plan - How to prepend a digit

2005-04-25 Thread Stiffe
Do you mean like this?

This is an example
Here I have 3 trunks (2 SIP and 1 IAX)

To dial out in rix, i'll have to dial a 0 before areaprefix, which
here always begins with another 0 (040xx)
 I.e  to dial that area on trunk rix  I'll have to dial 0040xxx = (00Z.)
EXTEN:1 will remove one (1) leading digit, the first 0 and I'm out on
that trunk.

Same with the digisip-trunk, instead I'll have to dial 9 to reach out
on that trunk
Same with the FWD-trunk, I'll have to dial 393 to reach the FWD network.
EXTEN:3 will remove the leading three digits (393)

To be able to dial another country I'll have to be able to dial 00 (000Z.)
Aka 0 00x  or 9 00x
ยด
I'm using [EMAIL PROTECTED] ver 0.9 with asterisk 1.0.7


;[outrt-001-rixroute]
;include = outrt-001-rixroute-custom
;exten = _000Z.,1,Macro(dialout-trunk,2,${EXTEN:1})
;exten = _000Z.,2,Macro(outisbusy) ; No available circuits
;exten = _00Z.,1,Macro(dialout-trunk,2,${EXTEN:1})
;exten = _00Z.,2,Macro(outisbusy)  ; No available circuits
;
;[outrt-002-digisiproute]
;include = outrt-002-digisiproute-custom
;exten = _900Z.,1,Macro(dialout-trunk,1,${EXTEN:1})
;exten = _900Z.,2,Macro(outisbusy) ; No available circuits
;exten = _90Z.,1,Macro(dialout-trunk,1,${EXTEN:1})
;exten = _90Z.,2,Macro(outisbusy)  ; No available circuits
;
;[outrt-003-fwdroute]
;include = outrt-003-fwdroute-custom
;exten = _393.,1,Macro(dialout-trunk,3,${EXTEN:3})
;exten = _393.,2,Macro(outisbusy)  ; No available circuits

Rgds

Stefan (Sweden)

On 4/25/05, Daniel Salama [EMAIL PROTECTED] wrote:
 I'd like to create a dial rule that when someone tries to dial a
 particular number, the same number is dialed, except that prefixed with
 some additional digit(s). How can this be specified on extensions.conf?
 
 Thanks,
 Daniel
 
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Asterisk2mp3

2005-04-24 Thread Stiffe
Hi folks.

I have a problem with a pythonscript designed for joining one in-wav
and one out-wav
after recording a call.

Yes, I have the wav-files after a successfull recording...

Python stops at line 5:basename = sys.argv[2]

like:

[EMAIL PROTECTED] bin]# python /usr/local/bin/asterisk2mp3.py
Traceback (most recent call last):
  File /usr/local/bin/asterisk2mp3.py, line 5, in ?
basename = sys.argv[2]
IndexError: list index out of range

What do I have to do?

And finally, can I have the script to mail the resulting mp3 to a mailadress
and after that delete the mp3-file, for saving space?

Thanks in advance

//Stefan (from Sweden)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[Asterisk-Users] Re: Asterisk2mp3

2005-04-24 Thread Stiffe
Sorry, I forgot to show the script, here it is:

#!/usr/bin/python
import tempfile,os,sys,re,time

monitordir = sys.argv[1]
basename = sys.argv[2]

def runcmd(cmd):
print cmd
os.system(cmd)

#mix to one wav
inwav = os.path.join(monitordir, basename+-in.wav)
outwav = os.path.join(monitordir, basename+-out.wav)
waste, mixedwav = tempfile.mkstemp(.wav,audiopipe_,/tmp)
runcmd (soxmix %s %s %s % (inwav,outwav,mixedwav))

#up sample rate
waste, uppedsamplerate = tempfile.mkstemp(.wav,audiopipe_,/tmp)
runcmd(sox %s -r 22050 %s  % (mixedwav, uppedsamplerate))

#run lame
outfile =  os.path.join(monitordir, basename+.mp3)

#I use gogo instead of lame, cause gogo is a lot faster. But if you
#can't compile gogo, lame will do just fine.
#
runcmd(lame -S -v %s %s % (uppedsamplerate, outfile))
#runcmd(/usr/local/bin/gogo -v 6  %s %s % (uppedsamplerate, outfile))





os.remove(inwav)
os.remove(outwav)

#but at least we can waste the temporaries
os.remove(mixedwav)
os.remove(uppedsamplerate)


On 4/24/05, Stiffe [EMAIL PROTECTED] wrote:
 Hi folks.
 
 I have a problem with a pythonscript designed for joining one in-wav
 and one out-wav
 after recording a call.
 
 Yes, I have the wav-files after a successfull recording...
 
 Python stops at line 5:basename = sys.argv[2]
 
 like:
 
 [EMAIL PROTECTED] bin]# python /usr/local/bin/asterisk2mp3.py
 Traceback (most recent call last):
  File /usr/local/bin/asterisk2mp3.py, line 5, in ?
basename = sys.argv[2]
 IndexError: list index out of range
 
 What do I have to do?
 
 And finally, can I have the script to mail the resulting mp3 to a mailadress
 and after that delete the mp3-file, for saving space?
 
 Thanks in advance
 
 //Stefan (from Sweden)

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users