Re: [asterisk-users] HFC-S card

2010-02-22 Thread Pedro Santos
On 2/22/2010 10:26 AM, Per Jessen wrote:
 Pedro Santos wrote:


 Does any one put a HFC-S card working in nt ptp mode?
  
 I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
 that helps.


 /Per Jessen, Zürich


I have use this howto
http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put 
the card working in nt ptp mode.
Can you explain me how i have to do that? Do you have any howto to make 
the card work in nt ptp mode?
thanks for answer

/Pedro Santos

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] HFC-S card

2010-02-22 Thread Pedro Santos
On 2/22/2010 7:36 AM, Tzafrir Cohen wrote:
 On Sun, Feb 21, 2010 at 07:55:39PM +, Pedro Santos wrote:

 Does any one put a HFC-S card working in nt ptp mode?
  
 Which version of Asterisk do you use? Which channel driver?


I have use this howto
http://www.voip-info.org/wiki/view/Asterisk+zaphfc;

Pedro Santos

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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


Re: [asterisk-users] HFC-S card

2010-02-22 Thread Pedro Santos
On 2/22/2010 1:02 PM, Tzafrir Cohen wrote:
 On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote:

 On 2/22/2010 10:26 AM, Per Jessen wrote:
  
 Pedro Santos wrote:



 Does any one put a HFC-S card working in nt ptp mode?

  
 I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
 that helps.


 /Per Jessen, Zürich



 I have use this howto
 http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put
 the card working in nt ptp mode.
 Can you explain me how i have to do that? Do you have any howto to make
 the card work in nt ptp mode?
 thanks for answer
  
 Short answert:   signalling = bri_net

 Longer answer:

 That page is outdated (hmm, and I didn't get to update it :-(   )

 Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The
 zaphfc driver, though, is still not included in DAHDI. It's maintained,
 though. The version included in the Debian packages is taken from
 http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary .

 Either way (bristuff or Asterisk= 1.6.0) to use BRI PTP NT in chan_dahdi
 you should set:

signalling = bri_net

 for the span's channels in /etc/asterisk/chan_dahdi.conf .


I´m using centos 4.8 server, and i don't now how integrate zaphfc with dadhi

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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


[asterisk-users] HFC-S card

2010-02-21 Thread Pedro Santos
Does any one put a HFC-S card working in nt ptp mode?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] automatic calls

2009-09-07 Thread Pedro Santos
hi,

can anyone knows a way to make automatic calls from a list of numbers stored
in a file, one by one, as the calls hangs up.

EX:

1º call - hang up - 2º call - hang up - 3º call ..


thanks,

pn
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Asterisk and Media gateway controller

2008-11-02 Thread Pedro G
Hello everyone, I am new in voip.

I want to use a linux pc as a media gateway controller (with Megaco
protrocol if possible). I heard Asterisk could do it, but in the
documentation I haven't found information about it.

Could someone help me?

Thank you very much.

Pedro Gonzalez

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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


[asterisk-users] alcatel omnipcx

2008-01-31 Thread Pedro Santos
Hi,

can anyone tell me how i do a sip trunk between an asterisk and a alcatel
omnipcx pbx with sip support

tx,

Pedro Santos
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] linksys spa3102 for faxing

2007-10-09 Thread pedro noticioso
Hi, I have been considering a purchase of the linksys spa3102 for a couple 
hours but I would like to know from someone here, wether this device will 
support faxing on my local asterisk server, I have had success sending and 
recieving faces with an x100p, and recall that in the old documentation, they 
mention that if I send/recieve faxes, that it all should be done on the local 
server for best performanc, so Im asuming tha this device may apply because 
there will be an ethernet cable between the FXO and the asterisk server?

thanks!




  

Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, 
and more!
http://tv.yahoo.com/collections/3658 

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

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


[asterisk-users] basic 3+ way conference call on plain old phones

2007-05-24 Thread pedro noticioso
hi guys, is it possible to do a basic 3-or-more-way
conference call when the phones dont support it? I am
fully aware of this concept on expensive phones like
this one:

Grandstream GXP 2000 -Conference call 3-way
http://www.youtube.com/watch?v=hlZ6JqE1MT4

The problem is that the basic plain old commercial PBX
supports 3-way calling in ugly old phones like this
one:

http://www.neo-shop.com/tiendas/0009/varios/telefono%20TEIDE-1.jpg

connected to an ata like this one:

http://www.egk.com.ar/imagenes/hardware/sipura2.jpg

The idea is to be caller (A): dial calle (B), once (B)
answers press on HOOK or something else to send them
to MOH, then dial callee (C), talk to him a little
too, then press the same HOOK or something else and
the 3, (A)(B) and (C) in a conference call.

Unlike the grandstream, this would definitelly have to
be done by *, isnt this part of the basic
functionality like voicemail that is already done and
a couple lines in the config files it will work on all
phones done by *?

if not, then, how do you recommend me to it? 

the closest I have seen to shat I am looking for is

http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro

is there a better alternative?

any thoughts?

thanks a lot!



   
Got
 a little couch potato? 
Check out fun summer activities for kids.
http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz
 
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] xten will not send tones to * and i from sip phone

2007-05-18 Thread pedro noticioso
hi there!

I have a couple phones connected to a sipura ata and
if I go into *- IVR, I press options on the regular
phones and it all works fine and dandy.

then I connect an xten softphone, a new extension in
my dialplan, I dial the ivr, * asks me to dial
something to go through it, I press keys on xten, but
nothing happens, * just times out through as if I did
not press anything!

is there some sort of configuration out there to tell
the xten softphone to work as expected? thanks!

Then another problem!

I used the i extension, plus _X and _X. to make sure I
catch everything that is not propperly dialed.

If I take the regular phones that are connected
through the sipura ata, then dial 'exten =
700,1,Goto(default,s,1)' so that I get the asking for
an extension to reach, I dial a wrong number and
walla, its caight by one of my magic numbers!

BUT, if I pickup the same phone, and just dial the
same wrong number? I just get a busy signal! and there
is nothing registered at the CLI even though I added
DEBIG to the configuration! :s

What can I do to make sure I always send an error
sound and never again a busy signal?


thanks!






 

Bored stiff? Loosen up... 
Download and play hundreds of games for free on Yahoo! Games.
http://games.yahoo.com/games/front
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] muscionhold error message

2007-05-11 Thread pedro noticioso
hi there guys!

how can I eliminate this message?

[May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
monmp3thread: Unable to spawn mp3player
[May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
spawn_mp3: Found no files in
'/var/lib/asterisk/mohmp3'

This is on debian etch 4.0
asterisk 1.4, it happens quite often everyday and I
have to scroll a lot to try to find other error
messages.

btw can I just put some musica wav files in
/var/lib/asterisk/mohmp3 ? that would be great to
leave asterisk's processor alone

thanks!


   
Luggage?
 GPS? Comic books? 
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] Sound files

2007-05-08 Thread Pedro Silva

Hello,

Can i identify the sound files that are played in the asterisk
console? I defined the verbose to 100 but i can not see the sound
files that are played in some situations... :(
For example, I need to know what files are played for the message:
Extension xxx is unavailable
The goal is to translate that to Portuguese (pt_pt)...

Thanks in advance,
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] Catch all undefined numbers to play a nice message and restart

2007-04-12 Thread pedro noticioso
Hi there list!

I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like

You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!



;                 
     
;
;
;
;
; begin extensions
;
;
;
;
;                 
     
;

[general]   ;

language=es
; autofallthrough=yes
clearglobalvars=no

[globals] 

; Definiendo variables para usarlas a traves de todo
el 
; MINOMBRE=mailinator.net
; MITELEFONOFXO=
; OPERADORA=



;
; Si static esta en no, u omitido, entonces pbx_config
va a sobreescribir
; a este archivo  cuando se cambien las extensiones.
Recuerda que todos los
; comentarios de este archivo desapareceran si pasa
eso.
;
; XXX Todavia no ha sido implementado XXX
;
static=yes
;
;
; si stati=yes y writeprotect=no, tambien puedes
guardar al dialplan con
; linea de comandos ejecutando 'save dialplan' y
borrando estos comentarios
;
writeprotect=yes

CONSOLE=Zap/1   ; pendiente entender *
TRUNK=Zap/1 ; Trunk interface *
TRUNKMSD=1  ; MSD digits to strip (usually
1 or 0) *



;                 
     
; Trunks

;[context] ;exten =
someexten,priority[+offset][(alias)],application(arg1,arg2,...)

[trunkint]  ; International long distance
through trunk
exten = _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld]   ; Long distance context
accessed through trunk
exten =
_901ZX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunklocal]; Local eight-digit dialing
accessed through trunk interface
exten =
_9ZXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})  ;
llamada local comun y corriente
exten = _90ZXS0,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
; 020, etc
exten = _9066,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
066, etc

[trunktollfree] ; Long distance context
accessed through trunk interface
exten =
_901800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkpaypercall] ; Dangerous pay-per call!
exten =
_901900.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkcelular]  ; Long distance context
accessed through trunk interface
exten =
_9044ZZ,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten =
_9045ZZ,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})



;                 
     
; Contexts
[international] ; Master context for
international long distance
ignorepat = 9
include = longdistance
include = trunkint

[longdistance]  ; Master context for long
distance
ignorepat = 9
include = local
include = trunkld
include = trunktollfree
include = trunkpaypercall

[mercadotecnia]
ignorepat = 9
include = local

[local] ; Master context for local,
toll-free, and iaxtel calls only
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal


[record]
exten = s,1,Answer
exten = s,2,Read(RECORD|enter4digits|4)
exten = s,3,Playback(record-instructions)
exten =
s,4,Record(/var/lib/asterisk/sounds/recording/s-${RECORD}|wav)
exten = s,5,Wait(2)
exten =
s,6,Playback(/var/lib/asterisk/sounds/recording/s-${RECORD})
exten = s,7,ResponseTimeout(10)
exten =
s,8,Background(1toaccept2torerecord3torecordanother)
exten = 1,1,Hangup
exten = 2,1,Goto(s,3)
exten = 3,1,Goto(s,2)


[macro-stdexten];
;
; Macro de extensiones estandard:
;   ${ARG1} - Extension  (Pudimos haver usado
${MACRO_EXTEN} tambien aqui
;   ${ARG2} - Aparato(s) a marcar
;
exten = s,1,Dial(${ARG2},20)   ; Ring the interface,
20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)  ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten = s-BUSY,1,Voicemail(${ARG1},b)  ; If busy,
send to voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten = _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything
else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain





[macro-stdexten-viejo] ; Standard extension macro:
; ARG1 es el numero de la extension
; ARG2 es sip al cual voy a marcar
exten = s,1,Dial(${ARG2},20,rt) ; Ring the interface,
20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If
they press #, return to start
exten = s-BUSY,1,Voicemail(b${ARG1})   ; If

[asterisk-users] Re: zapata with Tiger3XX compilation error

2007-03-15 Thread pedro noticioso
Ok so I read the Linux 2.6 related README and finally
compiled propperly, I thought but at the end I notice
that lscpi does report the cards, but I cant modprobe
wcfxo nor zaptel and I do have wcfxo.ko in the
/lib/modules/2.6.8/extra/ directory, so what gives? 

This is a Debian Sarge, thanks!





#
# make clean starts here
#
make[1]: Entering directory
`/usr/src/zaptel-1.4.0/menuselect'
rm -f menuselect *.o
make[2]: Entering directory
`/usr/src/zaptel-1.4.0/menuselect/mxml'
/bin/rm -f mxmldoc.o testmxml.o mxml-attr.o
mxml-entity.o mxml-file.o mxml-index.o mxml-node.o
mxml-search.o mxml-set.o mxml-private.o mxml-string.o
libmxml.a mxmldoc doc/mxml.3 doc/mxmldoc.1 testmxml
mxml.xml
/bin/rm -f mxmldoc-static libmxml.a
/bin/rm -f *.bck *.bak
/bin/rm -f config.cache config.log config.status
/bin/rm -f -r autom4te*.cache
make[2]: Leaving directory
`/usr/src/zaptel-1.4.0/menuselect/mxml'
make[1]: Leaving directory
`/usr/src/zaptel-1.4.0/menuselect'
rm -f torisatool makefw tor2fw.h radfw.h
rm -f fxotune fxstest sethdlc-new ztcfg ztdiag
ztmonitor ztspeed zttest zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f libtonezone.so libtonezone.a *.lo
make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.0
clean
make[1]: Entering directory
`/usr/src/kernel-source-2.6.8'
  CLEAN   /usr/src/zaptel-1.4.0/wct4xxp
  CLEAN   /usr/src/zaptel-1.4.0/.tmp_versions
make[1]: Leaving directory
`/usr/src/kernel-source-2.6.8'
rm -f xpp/*.ko xpp/*.mod.c xpp/.*o.cmd
rm -f xpp/*.o xpp/*.mod.o
rm -rf .tmp_versions
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f fxotune
rm -f core
rm -f ztcfg-shared fxstest
rm -rf misdn*
rm -rf mISDNuser*
#
# ./configure starts here
#
checking for gcc... gcc
checking for C compiler default output file name...
a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables... 
checking for suffix of object files... o
checking whether we are using the GNU C compiler...
yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none
needed
checking how to run the C preprocessor... gcc -E
checking for a BSD-compatible install...
/usr/bin/install -c
checking whether ln -s works... yes
checking for GNU make... make
checking for grep... /bin/grep
checking for sh... /bin/sh
checking for ln... /bin/ln
checking for grep that handles long lines and -e...
(cached) /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking for initscr in -lcurses... yes
checking curses.h usability... yes
checking curses.h presence... yes
checking for curses.h... yes
checking for initscr in -lncurses... yes
checking for curses.h... (cached) yes
checking for newtBell in -lnewt... yes
checking newt.h usability... yes
checking newt.h presence... yes
checking for newt.h... yes
checking for usb_init in -lusb... no
configure: creating ./config.status
config.status: creating build_tools/menuselect-deps
config.status: creating makeopts
configure: *** Zaptel build successfully configured
***
#
#  make linux26 starts here
#
make[1]: Entering directory
`/usr/src/zaptel-1.4.0/menuselect'
make[2]: Entering directory
`/usr/src/zaptel-1.4.0/menuselect'
make[3]: Entering directory
`/usr/src/zaptel-1.4.0/menuselect/mxml'
gcc -O -Wall   -c mxml-attr.c
gcc -O -Wall   -c mxml-entity.c
gcc -O -Wall   -c mxml-file.c
gcc -O -Wall   -c mxml-index.c
gcc -O -Wall   -c mxml-node.c
gcc -O -Wall   -c mxml-search.c
gcc -O -Wall   -c mxml-set.c
gcc -O -Wall   -c mxml-private.c
gcc -O -Wall   -c mxml-string.c
/bin/rm -f libmxml.a
/usr/bin/ar crvs libmxml.a mxml-attr.o mxml-entity.o
mxml-file.o mxml-index.o mxml-node.o mxml-search.o
mxml-set.o mxml-private.o mxml-string.o
a - mxml-attr.o
a - mxml-entity.o
a - mxml-file.o
a - mxml-index.o
a - mxml-node.o
a - mxml-search.o
a - mxml-set.o
a - mxml-private.o
a - mxml-string.o
ranlib libmxml.a
make[3]: Leaving directory
`/usr/src/zaptel-1.4.0/menuselect/mxml'
gcc -Wall  -o menuselect.o -g -c -D_GNU_SOURCE
menuselect.c
gcc -Wall  -o menuselect_curses.o -g -c -D_GNU_SOURCE 
menuselect_curses.c
gcc -Wall  -o strcompat.o -g -c -D_GNU_SOURCE
strcompat.c
gcc -g -Wall -o menuselect menuselect.o
menuselect_curses.o strcompat.o mxml/libmxml.a
-lncurses 
make[2]: Leaving directory
`/usr/src/zaptel-1.4.0/menuselect'
make[1]: Leaving directory
`/usr/src/zaptel-1.4.0/menuselect'
gcc gendigits.c  -lm -o gendigits
./gendigits  tones.h
gcc -o makefw makefw.c
./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw  radfw.h
Loaded 42096 bytes from file
make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.4.0
modules
make[1]: Entering directory
`/usr/src/kernel-source-2.6.8'
  CC [M]  

[asterisk-users] asterisk

2007-02-23 Thread Pedro Santos

Hi

i install Asterisk can register softphones on clients computers but when i
make a call to a extencion this error apear
Call Failed: not found

in the asterisk machine i do commannd sip show peers and i can see the
clients there

can you help me

thanks
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Freepbx changes dont reflect in asterisk

2006-11-18 Thread Pedro Silva

Alex was right. The problem is that when i make changes in freepbx,
those changes are not written in the config files.
I only made modifications in files_custom.conf.

The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1.

Thanks by your help,
Ps.


2006/11/18, Alex Robar [EMAIL PROTECTED]:

I think you guys are all misunderstanding the problem here. Unless I'm
misunderstanding, Pedro's issue is that when he makes changes in FreePBX,
those changes are not written out to the config files.

Alex

 On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
 You can't do any modifications in extensions_additional.conf and
sip_additional.conf. Same is true for extensions.conf and sip.conf, and
other original trixbox files. As soon as you press the red bar, they are
returned to their original state. For modifications, you create your own
files or use sip_customs.conf and extensions_custom.conf.

 Please don't mix trixbox with asterisk just because its based on asterisk.
Its a completely customized solution of various software packages configured
to make it work according to its own requirements. For help, post on
trixbox.org forums.

 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users






--
Alex Robar
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Freepbx changes dont reflect in asterisk

2006-11-18 Thread Pedro Silva

I also restarted the box and the problem is not solved :(
PS

2006/11/18, Dumpolid Exeplish [EMAIL PROTECTED]:

i also used to have this problem, for instance we use the pin code
functionality of FreePBX and whenever i add or modify a pin number, it is
not effected or changed in the config files. i dont know what causes this
error but i have noticed that restarting FreePBX or re-installing the
application stops this. Just restart the box



On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote:
 Alex was right. The problem is that when i make changes in freepbx,
 those changes are not written in the config files.
 I only made modifications in files_custom.conf.

 The version of freePbx that i use is 2.1.1 (not beta) and Asterisk
1.2.12.1.

 Thanks by your help,
 Ps.


 2006/11/18, Alex Robar [EMAIL PROTECTED]:
  I think you guys are all misunderstanding the problem here. Unless I'm
  misunderstanding, Pedro's issue is that when he makes changes in
FreePBX,
  those changes are not written out to the config files.
 
  Alex
 
   On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
   You can't do any modifications in extensions_additional.conf and
  sip_additional.conf. Same is true for extensions.conf and sip.conf, and
  other original trixbox files. As soon as you press the red bar, they are
  returned to their original state. For modifications, you create your own
  files or use sip_customs.conf and extensions_custom.conf.
  
   Please don't mix trixbox with asterisk just because its based on
asterisk.
  Its a completely customized solution of various software packages
configured
  to make it work according to its own requirements. For help, post on
  trixbox.org forums.
  
   ___
   --Bandwidth and Colocation provided by Easynews.com --
  
   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit:
  
 
http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  
 
 
 
  --
  Alex Robar
  [EMAIL PROTECTED]
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 
 
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] Freepbx changes dont reflect in asterisk

2006-11-17 Thread Pedro Silva

Hello,


From some days ago, when i made changes in web interface to asterisk

that comes with trixbox (freepbx), this dont reflect the changes in
asterisk configuration.
I has reviewed the file permissions in /etc/asterisk and all files are
writable to asterisk user.
In freepbx all appears to be ok (i dont see any errors...).

Anyone can help me with this problem?
Thanks in advance,
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Freepbx changes dont reflect in asterisk

2006-11-17 Thread Pedro Silva

Hi,

2006/11/17, Alex Robar [EMAIL PROTECTED]:

Hi Pedro,

Did you press the red bar at the top of the page? Until you do this, the
config files are not written out.


Yes, i press the red bar and freepbx dont return any error.
For example, If i add a new extension, the files
extensions_addicional.conf and sip_addicional.con are supposed to be
updated and are not.

Best regards,
PS.



Alex


On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote:

 Hello,

 From some days ago, when i made changes in web interface to asterisk
 that comes with trixbox (freepbx), this dont reflect the changes in
 asterisk configuration.
 I has reviewed the file permissions in /etc/asterisk and all files are
 writable to asterisk user.
 In freepbx all appears to be ok (i dont see any errors...).

 Anyone can help me with this problem?
 Thanks in advance,
 PS.
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




--
Alex Robar
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] jpeglib

2006-11-08 Thread Pedro Silva

Hello,

When i try to install the sfftobmp3.1, the tribbox box give me the
following error:
...
checking for TIFFOpen in -ltiff... yes
checking jpeglib.h usability... no
checking jpeglib.h presence... no
checking for jpeglib.h... no
configure: error: jpeglib.h not found

I try to find packages with jpeglib but i cannot find that... :(
Someone can tell me where i can find that package?

Thanks in advance!
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] capiAnswerFax

2006-11-07 Thread Pedro Silva

Hello,

Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten = 1,1,Goto(handle_fax,s,1)
exten = fax,1,Goto(handle_fax,s,1)

[handle_fax]
exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten = s,2,Hangup()

With this code, a fax call to DID 1 must be attended and the fax
stored in /tmp, right?
This not works... :(

Thanks for any kind of possible help...
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] capiAnswerFax

2006-11-07 Thread Pedro Silva

Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.

2006/11/7, Michiel van Baak [EMAIL PROTECTED]:

On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
 Hello,

 Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
 I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
 it give me errors...
 Also i tried define one extension for fax receptions but this dont works:
 exten = 1,1,Goto(handle_fax,s,1)
 exten = fax,1,Goto(handle_fax,s,1)

 [handle_fax]
 exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
 exten = s,2,Hangup()

 With this code, a fax call to DID 1 must be attended and the fax
 stored in /tmp, right?
 This not works... :(

 Thanks for any kind of possible help...
 PS.

Hi,

The chan_capi you mention already has fax support.
Here is the handle_fax context I use with the latest
released chan_capi-cm

[handle_fax]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten = s,n,capicommand(receivefax|${FAXFILE})
exten = h,1,DeadAgi(faxreceive.php|${FAXFILE})

Good luck
--

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-11-02 Thread Pedro Silva

2006/11/1, Armin Schindler [EMAIL PROTECTED]:

On Wed, 1 Nov 2006, Pedro Silva wrote:



As you can see in the log below, the called number is just '0':
 CalledPartyNumber   = 810

It seems DDI 0 of your line was called. So just do
  exten = 0,n,Dial...

Armin


Is that right! Thanks!
Best regards,
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] How to clear trixbox configuration

2006-11-02 Thread Pedro Silva

Hello all,

To test some configs i forgot the trixbox web config (freepbx) and i
made changes directly in asterisk config files (sip.conf,
extensions.conf, etc). Result: asterisk is working ok but the the web
config is totaly confused and, if i made a change via freepbx this not
works ok. Only now i know that this changes will be made in
file_custom.conf... problem of newbie... :).
I also updated the asterisk for version 1.2.12.1, independently for
the trixbox updating system. My trixbox version is 1.2.2.
So i need to clear all configuration and start again only with the web
config in freepbx.
Is possible to clear all web configs and restitute all initial
/etc/asterisk/* files to start from zero without re-installing all
trixbox box from CD?

Thanks in advance!
Best regards,
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-11-01 Thread Pedro Silva

Hello,

The problem was wrong contexts defined like Marco said, and is solved.
Now, i have another problem...of course :)

On incoming calls, i only can receive calls if i define a line like
the following, in extensions.conf:
exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected
to extension 500).
The problem is that i have some DDI's assigned by my telco (xxx302500
to xxx302509) and i need to route each DDI to diferent internal
extension.
If i define someting like exten = _0,n,Dial... (for DDI
xxx302500) the call is not answered by asterisk. I think that asterisk
cannot identify the destination DDI of the incoming call...is this
normal?
This is the capi debug of one incoming call:

asterisk1*CLI
CONNECT_IND ID=001 #0x1975 LEN=0045
 Controller/PLCI/NCCI= 0x401
 CIPValue= 0x10
 CalledPartyNumber   = 810
 CallingPartyNumber  = 00 83X
 CalledPartySubaddress   = default
 CallingPartySubaddress  = default
 BC  = 80 90 a3
 LLC = default
 HLC = 91 81
 AdditionalInfo
  BChannelinformation= default
  Keypadfacility = default
  Useruserdata   = default
  Facilitydataarray  = default

   -- CONNECT_IND (PLCI=0x401,DID=0,CID=X,CIP=0x10,CONTROLLER=0x1)
   ISDN1#02: msn='*' DNID='0' MSN
 == ISDN1#02: setting format alaw - 0x8 (alaw)
 == ISDN1#02: Incoming call 'X' - '0'
INFO_IND ID=001 #0x1976 LEN=0017
 Controller/PLCI/NCCI= 0x401
 InfoNumber  = 0x70
 InfoElement = 810

INFO_RESP ID=001 #0x1976 LEN=0012
 Controller/PLCI/NCCI= 0x401

   -- ISDN1#02: info element CALLED PARTY NUMBER
   ISDN1#02: INFO_IND DID digits not used in this state.
INFO_IND ID=001 #0x1977 LEN=0015
 Controller/PLCI/NCCI= 0x401
 InfoNumber  = 0xa1
 InfoElement = default

INFO_RESP ID=001 #0x1977 LEN=0012
 Controller/PLCI/NCCI= 0x401

   -- ISDN1#02: info element Sending Complete
CONNECT_RESP ID=001 #0x1977 LEN=0032
 Controller/PLCI/NCCI= 0x401
 Reject  = 0x1
 BProtocol
  B1protocol = 0x0
  B2protocol = 0x0
  B3protocol = 0x0
  B1configuration= default
  B2configuration= default
  B3configuration= default
 ConnectedNumber = default
 ConnectedSubaddress = default
 LLC = default
AdditionalInfo
  BChannelinformation= default
  Keypadfacility = default
  Useruserdata   = default
  Facilitydataarray  = default

INFO_IND ID=001 #0x1978 LEN=0016
 Controller/PLCI/NCCI= 0x401
 InfoNumber  = 0x18
 InfoElement = 81

INFO_RESP ID=001 #0x1978 LEN=0012
 Controller/PLCI/NCCI= 0x401

   -- ISDN1#02: info element CHANNEL IDENTIFICATION 81
INFO_IND ID=001 #0x1979 LEN=0015
 Controller/PLCI/NCCI= 0x401
 InfoNumber  = 0x8005
 InfoElement = default

INFO_RESP ID=001 #0x1979 LEN=0012
 Controller/PLCI/NCCI= 0x401

   -- ISDN1#02: info element SETUP
   ISDN1#02: IE SETUP / SENDING-COMPLETE already received.
DISCONNECT_IND ID=001 #0x197b LEN=0014
 Controller/PLCI/NCCI= 0x401
 Reason  = 0x0

DISCONNECT_RESP ID=001 #0x197b LEN=0012
 Controller/PLCI/NCCI= 0x401

   -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup.
   CAPI/ISDN1/0-15: set channel task to 1
 == ISDN1#02: CAPI Hangingup for PLCI=0x401 in state 4
 == ISDN1#02: Interface cleanup PLCI=0x401
   CAPI devicestate requested for ISDN1/0

Anyone can give me ideas about this problem?
Thanks in advance!
Best regards,
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-10-30 Thread Pedro Silva

Hello,

One problem is solved and another appears... :(
I cannot receive incoming calls on trixbox. I defined one incoming
route (any DID/any CID) and forwading these calls to a SIP extension.
With capi and sip debug in asterisk -r console i dont detect any
incoming activity...
In xlog console i have the following debug:
 0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B
00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81
Q.931  CR0d SETUP
   Sending complete
   Bearer Capability 80 90 a3
   Channel Id 81
   Calling Party Number 00 83 '963045723'
   Called Party Number 81 '0'
   HLC 91 81
   0:1898:127 - SIG-S 0-6 e:805
   0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec
   0:1898:130 - alloc cr in use =4
   0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95
Q.931  CR8d DISC
   Cause 80 95 'Call rejected'
   0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8
Q.931  CR8d REL_COM
   Cause 80 d8 'Incompatible destination'
   0:1898:133 - SIG-S 6-0 e:8c5
   0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8
   0:1898:135 - free cr in use =3
   0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec
   0:1898:155 - D-R(004) 00 01 01 16

So the problem appears to be Incompatible destination... but is
problem in asterisk or is before asterisk, on diva card...?

Tanks by any possible help!
Best regards,
PS.

2006/10/29, Pedro Silva [EMAIL PROTECTED]:

Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port... :(
- the card config needs to be: ETSI; TE; Point-to-Point (I thought
that was point-to-multipoint).

Best regards,
PS.

2006/10/29, Pedro Silva [EMAIL PROTECTED]:
 Hello again Alberto!

  Anyway, to get more info, try to open a second shell
  and run /usr/lib/eicon/divas/xlog
  then on the first shell redo the telsampl test, then
  post the output of xlog off the list to my address
  (alberto at msoft-italia.com)

 This is the xlog output:
 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
 4:1736:074 - alloc cr in use =4
 4:1736:076 - free cr in use =3
 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb
 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
 4:1736:078 - alloc cr in use =4
 4:1736:080 - free cr in use =3
 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb
 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
 4:1736:081 - alloc cr in use =4
 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048
 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08
 4:1736:084 - [1,0] Download 532 requested
 4:1736:084 - MORE
 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E
 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33
  Q.931  CR36 SETUP
 Sending complete
 Bearer Capability 80 90 a3
 Channel Id 83
 Progress Indicator 80 83
 Called Party Number 80 '963045723'
 4:1736:085 - SIG-S 0-1 e:885
 4:1736:087 - ACTIVATION_REQ
 4:1744:147 - L1_DOWN
 4:1744:150 - SIG-EVENT  08

 4:1744:150 - SIG-EVENT  08

 4:1744:150 - EVENT: Call failed in State 'Call initiated'
  Link disconnected, Layer-1 error (cable or NT)
 4:1744:150 - SIG-S 1-0 e:
 4:1744:151 - [1,0] dsp_release
 4:1744:155 - free cr in use =3
 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb

 I disconnect the rj45 cable from alcatel pbx and connect that to the
 diva card (with alcatel pbx i can make calls normally). The green led
 of the diva card is activated when i connect the cable. So i dont
 understand why the message  Link disconnected, Layer-1 error (cable
 or NT)...
 This debug is th same if the cable is connected to the NT or not.
 Any ideas...? Thanks!
 PS.



___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-10-29 Thread Pedro Silva

Thanks Alberto!

I tested with telsampl like you said (with various configurations for
de diva) and this not works...:(
The trace is:
Enter destination address: 273xx
--Conn_Req(273xx)
Connect_Con--
[29]:Disc_Ind--
--Disc_Res
**Call cleared***

Any idea for the possible problem?
Thanks and best regards,
PS.

2006/10/29, Alberto Pastore [EMAIL PROTECTED]:

Pedro Silva ha scritto:
 Hello,

 I need to connect one diva server 4bri to a portuguese BRI interface.
 The operator (PT) said that this bri is in point-to-multipoint mode
 (S0). Previously one PBX has connected to that interface.
 The asterisk and diva drivers are working ok but i cannot communicate
 to outside via this bri. Xlite gives me the message: call failed:
 declined.
 Anyone have experience with this setup?
 What are the main parameters for bri card configuration?
 D-channel protocol: ETSI-DSS1 or other?
 Interface mode: NT or TE?
 Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?)

 Thanks by any kind of help!
 Best regards,
 PS.
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
I'm not sure about Portuguese operators standard, but I bet
ETSI-DSS1 should work just fine. The interface mode is surely
TE.
The DID/MSN should not affect outgoing calls, I generally leave DID
off unless the telco company has that service active.

If you're using the diva server for linux package from eicon
(divas4linux, currently rel. 8.2), you should find a very
simple utility named telsampl under /usr/lib/eicon/divas
which you can run besides asterisk, to test outgoing calls.

You should run it with this command line: telsampl -c x
where x is the bri port you wish to test (1..4)
then at the prompt type c and enter a pstn number, e.g.
your mobile phone, then you can watch the log onscreen.

If the outgoing call works, then your isdn setup is correct,
and the problem is in asterisk. The message from xlite is not
meaningful, as it could occur on many situations.
You should watch the debug output on asterisk console.

That helped me a lot.

Alberto.


--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-10-29 Thread Pedro Silva

Olá Marco! :)

2006/10/29, Marco Mouta [EMAIL PROTECTED]:

pls post your misdn.conf as well as extensions.conf


The asterisk version that i used with trixbox dont't have
misdn.conf... I used capi.conf.
For now, i dont care about asterisk, because with the divas utility
telsampl i know that the problem is between diva card and BRI access.
So i need to solve first this problem and only after that im care with
asterisk... :)

Obrigado desde já pela disponibilidade de ajuda!
PS.



May be i can help.

Sou Português:)


On 10/29/06, Pedro Silva  [EMAIL PROTECTED] wrote:
 Thanks Alberto!

 I tested with telsampl like you said (with various configurations for
 de diva) and this not works...:(
 The trace is:
 Enter destination address: 273xx
 --Conn_Req(273xx)
 Connect_Con--
 [29]:Disc_Ind--
 --Disc_Res
 **Call cleared***

 Any idea for the possible problem?
 Thanks and best regards,
 PS.

 2006/10/29, Alberto Pastore [EMAIL PROTECTED]:
  Pedro Silva ha scritto:
   Hello,
  
   I need to connect one diva server 4bri to a portuguese BRI interface.
   The operator (PT) said that this bri is in point-to-multipoint mode
   (S0). Previously one PBX has connected to that interface.
   The asterisk and diva drivers are working ok but i cannot communicate
   to outside via this bri. Xlite gives me the message: call failed:
   declined.
   Anyone have experience with this setup?
   What are the main parameters for bri card configuration?
   D-channel protocol: ETSI-DSS1 or other?
   Interface mode: NT or TE?
   Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?)
  
   Thanks by any kind of help!
   Best regards,
   PS.
   ___
   --Bandwidth and Colocation provided by Easynews.com --
  
   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit:
  
http://lists.digium.com/mailman/listinfo/asterisk-users
  I'm not sure about Portuguese operators standard, but I bet
  ETSI-DSS1 should work just fine. The interface mode is surely
  TE.
  The DID/MSN should not affect outgoing calls, I generally leave DID
  off unless the telco company has that service active.
 
  If you're using the diva server for linux package from eicon
  (divas4linux, currently rel. 8.2), you should find a very
  simple utility named telsampl under /usr/lib/eicon/divas
  which you can run besides asterisk, to test outgoing calls.
 
  You should run it with this command line: telsampl -c x
  where x is the bri port you wish to test (1..4)
  then at the prompt type c and enter a pstn number, e.g.
  your mobile phone, then you can watch the log onscreen.
 
  If the outgoing call works, then your isdn setup is correct,
  and the problem is in asterisk. The message from xlite is not
  meaningful, as it could occur on many situations.
  You should watch the debug output on asterisk console.
 
  That helped me a lot.
 
  Alberto.
 
 
  --
  --
  Alberto Pastore
  B-Press Srl - Gruppo MSoft
  P.IVA 01697420030
  P.le Lombardia, 4 - 28100 Novara - Italy
  Tel. 0321-499508
  Fax 0321-492974
  http://www.msoft.it
 
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




 --
Com os melhores cumprimentos,

Marco Mouta
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-10-29 Thread Pedro Silva

Hello again Alberto!


Anyway, to get more info, try to open a second shell
and run /usr/lib/eicon/divas/xlog
then on the first shell redo the telsampl test, then
post the output of xlog off the list to my address
(alberto at msoft-italia.com)


This is the xlog output:
4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:074 - alloc cr in use =4
   4:1736:076 - free cr in use =3
   4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb
   4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:078 - alloc cr in use =4
   4:1736:080 - free cr in use =3
   4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb
   4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
   4:1736:081 - alloc cr in use =4
   4:1736:083 - [1,0] dsp_assign 0016, 0, 2048
   4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08
   4:1736:084 - [1,0] Download 532 requested
   4:1736:084 - MORE
   4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E
02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33
Q.931  CR36 SETUP
   Sending complete
   Bearer Capability 80 90 a3
   Channel Id 83
   Progress Indicator 80 83
   Called Party Number 80 '963045723'
   4:1736:085 - SIG-S 0-1 e:885
   4:1736:087 - ACTIVATION_REQ
   4:1744:147 - L1_DOWN
   4:1744:150 - SIG-EVENT  08

   4:1744:150 - SIG-EVENT  08

   4:1744:150 - EVENT: Call failed in State 'Call initiated'
Link disconnected, Layer-1 error (cable or NT)
   4:1744:150 - SIG-S 1-0 e:
   4:1744:151 - [1,0] dsp_release
   4:1744:155 - free cr in use =3
   4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb

I disconnect the rj45 cable from alcatel pbx and connect that to the
diva card (with alcatel pbx i can make calls normally). The green led
of the diva card is activated when i connect the cable. So i dont
understand why the message  Link disconnected, Layer-1 error (cable
or NT)...
This debug is th same if the cable is connected to the NT or not.
Any ideas...? Thanks!
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [asterisk-users] Diva server 4bri and Portuguese BRI

2006-10-29 Thread Pedro Silva

Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port... :(
- the card config needs to be: ETSI; TE; Point-to-Point (I thought
that was point-to-multipoint).

Best regards,
PS.

2006/10/29, Pedro Silva [EMAIL PROTECTED]:

Hello again Alberto!

 Anyway, to get more info, try to open a second shell
 and run /usr/lib/eicon/divas/xlog
 then on the first shell redo the telsampl test, then
 post the output of xlog off the list to my address
 (alberto at msoft-italia.com)

This is the xlog output:
4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
4:1736:074 - alloc cr in use =4
4:1736:076 - free cr in use =3
4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb
4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
4:1736:078 - alloc cr in use =4
4:1736:080 - free cr in use =3
4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb
4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb
4:1736:081 - alloc cr in use =4
4:1736:083 - [1,0] dsp_assign 0016, 0, 2048
4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08
4:1736:084 - [1,0] Download 532 requested
4:1736:084 - MORE
4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E
02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33
 Q.931  CR36 SETUP
Sending complete
Bearer Capability 80 90 a3
Channel Id 83
Progress Indicator 80 83
Called Party Number 80 '963045723'
4:1736:085 - SIG-S 0-1 e:885
4:1736:087 - ACTIVATION_REQ
4:1744:147 - L1_DOWN
4:1744:150 - SIG-EVENT  08

4:1744:150 - SIG-EVENT  08

4:1744:150 - EVENT: Call failed in State 'Call initiated'
 Link disconnected, Layer-1 error (cable or NT)
4:1744:150 - SIG-S 1-0 e:
4:1744:151 - [1,0] dsp_release
4:1744:155 - free cr in use =3
4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb

I disconnect the rj45 cable from alcatel pbx and connect that to the
diva card (with alcatel pbx i can make calls normally). The green led
of the diva card is activated when i connect the cable. So i dont
understand why the message  Link disconnected, Layer-1 error (cable
or NT)...
This debug is th same if the cable is connected to the NT or not.
Any ideas...? Thanks!
PS.


___
--Bandwidth and Colocation provided by Easynews.com --

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


[asterisk-users] Diva server 4bri and Portuguese BRI

2006-10-28 Thread Pedro Silva

Hello,

I need to connect one diva server 4bri to a portuguese BRI interface.
The operator (PT) said that this bri is in point-to-multipoint mode
(S0). Previously one PBX has connected to that interface.
The asterisk and diva drivers are working ok but i cannot communicate
to outside via this bri. Xlite gives me the message: call failed:
declined.
Anyone have experience with this setup?
What are the main parameters for bri card configuration?
D-channel protocol: ETSI-DSS1 or other?
Interface mode: NT or TE?
Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?)

Thanks by any kind of help!
Best regards,
PS.
___
--Bandwidth and Colocation provided by Easynews.com --

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


[Asterisk-Users] hi guys, a new newbie here needing help :D

2006-05-12 Thread pedro noticioso
I just installed rpm binaries in a new mandriva and I
see a frew error messages with asterisk -vvvcfg,
btw I would also like a little guidance to just set up
a couple sip phones to start playing with soft phone
communication with 3 pcs on the network

thanks :)


ng '/etc/asterisk/agents.conf': Found
 [skipping chan_alsa.so]
 [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
  == Registered custom function IAXPEER
May 12 15:50:12 WARNING[6173]: chan_iax2.c:9212
load_module: Unable to open IAX timing interface: No
such file or directory
  == Registered application 'IAX2Provision'
  == Manager registered action IAXpeers
  == Manager registered action IAXnetstats
  == Parsing '/etc/asterisk/iax.conf': Found
-- doing lookup for '216.207.245.47'
  == Registered channel type 'IAX2' (Inter Asterisk
eXchange Driver (Ver 2))
  == Using TOS bits 16
  == Binding IAX2 to default address 0.0.0.0:4569
  == IAX Ready and Listening
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
 [chan_local.so] = (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy
Channel Driver)
 [chan_mgcp.so] = (Media Gateway Control Protocol
(MGCP))
  == Parsing '/etc/asterisk/mgcp.conf






ng '/etc/asterisk/agents.conf': Found
 [skipping chan_alsa.so]
 [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
  == Registered custom function IAXPEER
May 12 15:50:12 WARNING[6173]: chan_iax2.c:9212
load_module: Unable to open IAX timing interface: No
such file or directory
  == Registered application 'IAX2Provision'
  == Manager registered action IAXpeers
  == Manager registered action IAXnetstats
  == Parsing '/etc/asterisk/iax.conf': Found
-- doing lookup for '216.207.245.47'
  == Registered channel type 'IAX2' (Inter Asterisk
eXchange Driver (Ver 2))
  == Using TOS bits 16
  == Binding IAX2 to default address 0.0.0.0:4569
  == IAX Ready and Listening
  == Loaded firmware 'iaxy.bin'
  == Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
 [chan_local.so] = (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy
Channel Driver)
 [chan_mgcp.so] = (Media Gateway Control Protocol
(MGCP))
  == Parsing '/etc/asterisk/mgcp.conf


 [codec_gsm.so] = (GSM/PCM16 (signed linear) Codec
Translator)
  == Parsing '/etc/asterisk/codecs.conf': Found
-- codec_gsm: using generic PLC
  == Registered translator 'gsmtolin' from format gsm
to slin, cost 1
May 12 15:50:34 WARNING[6173]: config_old.c:28
ast_load: ast_load is deprecated, use ast_config_load
instead!
  == Parsing '/etc/asterisk/rpt.conf': Found
  == Registered translator 'lintogsm' from format slin
to gsm, cost 3

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

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


[Asterisk-Users] Realtime goto problem

2006-04-18 Thread Pedro Nunes


Hi,


Sample database
++---+---+--+-+-

-+

| id | context   | exten | priority | app | appdata
|

++---+---+--+-+-

-+

|  1 | incoming| 6069  |1 | Goto|
incoming-next|6069|1 |

|  2 | incoming| 6069  |2 | Hangup  |
|

|  3 | incoming-next | 6069  |1 | DigitTimeout| 10
|

|  4 | incoming-next | 6069  |2 | ResponseTimeout | 30
|

|  5 | incoming-next | 6069  |3 | Background  | welcome


If i dont declare the incoming-next context in extensions.conf I get:
Channel 'Zap/21-1' sent into invalid extension '1' in context
'incoming-next ', but no invalid handler.

But if I put on extensions.conf:
[incoming-next]
Switch = Realtime/@
,it works fine.

Do we need to declare all contexts in extensions.conf so we can use it
on Realtime??

Another question:
Its possible to include contexts in Realtime like we made on
extensions.conf?


Thanks in advance,

Pedro Nunes


___
--Bandwidth and Colocation provided by Easynews.com --

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


[Asterisk-Users] H.323 ( HW PBX to *)

2006-02-28 Thread Pedro Mansilla








Hi,



 Im
trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with
other PBX. The port use to connect is TCP 1720 but I cant configure this
port on my * box. Im using a H.323.conf file sample to activate the port
but the * isnt listening there. Somebody have any idea or tip?



This is mi H.323.conf









[general]

port = 1720

bindaddr = 192.168.0.200

;tos=lowdelay

;



;

amaflags = default

;





;

;accountcode=lss0101

;





;

allow=all
; turns on all installed codecs

;disallow=g723.1
; Hm... Proprietary, don't use it...

;allow=gsm
; Always allow GSM, it's cool :)

;allow=ulaw

; User-Input Mode ( DTMF )

;

; valid entries are: rfc2833, inband

; default is rfc2833

dtmfmode=rfc2833

;

; Set the gatekeeper

;
DISCOVER
- Find the Gk address using multicast

;
DISABLE
- Disable the use of a GK

; IP address or Host name - The acutal
IP address or hostname of your GK

;gatekeeper = DISABLE

;



Tell Asterisk whether or not to accept Gatekeeper routed
calls or not. Normally

this should always be set to yes, unless you want to have
finer control over wh

ch users are allowed access to Asterisk. Default: YES



;

AllowGKRouted = yes

;



Default context gets used in siutations where you are using
the GK routed model

or no type=user was found. This gives you the ability to
either play an invalid



message or to simply not use user
authentication at all.







Thanks in advance.



Pedro.








___
--Bandwidth and Colocation provided by Easynews.com --

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


RE: [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC)

2006-02-09 Thread Pedro Nunes

Hi Sven,

Same problem... Not solved...
With CAPI and mISDN. 

I think it as to do with 

nationalprefix=0
internationalprefix=00

on capi.conf/misdn.conf. I already try to nationalprefix= but always
get that damn 0. If I change nationalprefix=5 I get a leading 5 and so
on... But without any leading digit I couldn't do it yet.

Pedro Nunes


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven
Fischer
Sent: quinta-feira, 9 de Fevereiro de 2006 9:23
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC)

Hi all,

I have a problem: On my internal S0 where phones are connected via HFC I
get 
all the number with a leading 0 (either from internal SIP phones or
external 
dialins via CAPI). I don't know where to look for this 0. Any ideas?

Greetings, Sven

-- 
Sven Fischer (Dipl.-Phys.) - FACIT Consulting GmbH
  Hausinger Str. 6 - 40764 Langenfeld
  Tel: 02173/16700-55 Fax: 02173/16700-60

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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


[Asterisk-Users] mISDN errors on asterisk CLI

2006-01-30 Thread Pedro Nunes

Hi there guys,

Does anyone know what this is??
Every time a mISDN channel connects to anything, I get this message on
the CLI of asterisk. 

Unhandled Message: prim 281 len -22 from addr 51400101, dinfo 0 on port:
1

Thanks 


Pedro Nunes
___
--Bandwidth and Colocation provided by Easynews.com --

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


RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold

2005-12-15 Thread Pedro Nunes
Hello,

Do you try

Answer() and then Dial(SIP/xyz,,m)???

Exten = ???,1,Answer()
Exten = ???,2,Dial(SIP/xyz,,m)

You need to answer the call before you can hear music on hold.

Pedro Nunes

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Clauson
Sent: quinta-feira, 15 de Dezembro de 2005 4:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Starting RTP with Dial and MusicOnHold

Hi,

I'm trying to get Asterisk working with a supplier's Cerpack switch and
everything is working except audio ringback for calls coming from
Cerpack to
Asterisk.

The Cerpack switch only does out of band progress indication (seems a
bit
strange for SIP to SIP calls?!) so I've spent the last two days trying
to
find a way to force Asterisk to send an RTP stream to Cerpack for ring
back.

Theoretically the Dial command with the m option looks to be exactly
what I
need:

Dial(SIP/xyz,,m)

This should play musiconhold back to the caller and in my case I just
took a
recording of the PSTN tones I wanted to play and created a musiconhold
class
for them. The command will work correctly when dialled from a SIP phone
connected to Asterisk but not for calls coming from Cerpack. As far as I
can
tell this is because Asterisk won't initiate the RTP stream and waits
for a
packet from the client before starting to play the musiconhold, perhaps
assuming the connection is not available until it gets a packet. In this
case Cerpack isn't sending a packet so no audio is heard until the call
is
answered.

Has anybody seen anything like this before?

Thanks,

Aaron


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

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


[Asterisk-Users] mISDN Caller ID problem

2005-12-13 Thread Pedro Nunes


Hello everyone,

I am trying mISDN driver with asterisk 1.2.1 but when i call from SIP to
mISDN and from mISDN to SIP, the caller ID appears always with a leading
0 (0X). I think the problem is with nationalprefix.

How can I remove that zero

Here is my config.

[general]

debug=0
trace_calls=false
trace_dir=/var/log/
bridging=yes
stop_tone_after_first_digit=yes
append_digits2exten=yes
l1_info_ok=yes
clear_l3=no
method=standard
dynamic_crypt=no

crypt_prefix=**
crypt_keys=test,muh

[default]
context=default
language=en
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
dialplan=0
use_callingpres=yes

;always_immediate=no
;immediate=no
;hold_allowed=yes
;callgroup=1
;pickupgroup=1
;presentation=not_screened
;echocancel=no
echocancelwhenbridged=no
echotraining=yes

[group1]
ports=1
context=bri_card_1
msns=*


Thanks in advance

Pedro Nunes
___
--Bandwidth and Colocation provided by Easynews.com --

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


[Asterisk-Users] Need advice on BRI

2005-12-12 Thread Pedro Nunes
Hello all,

I need to install a production server with BRI support.
I know that exists bristuff, misdn, chan_capi ...

I have hcfpci based cards.

For a very stable environment, what driver should I use??

Thanks in advance

Pedro Nunes

___
--Bandwidth and Colocation provided by Easynews.com --

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


Re: [Asterisk-Users] Codec that quality does not get affect *much* against packet loss

2005-11-22 Thread Pedro

I think you are thinking of iLBC:

http://www.voip-info.org/wiki-iLBC

Be aware that this codec is known to be pretty CPU intensive to accomplish its compression.

- PedroOn 11/22/05, Sam Tam [EMAIL PROTECTED] wrote:
I think I have heard in the past that someone mentioned to me there is acodec that does not getting affected much because of packet loss.Is there such thing?Sam___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Asterisk 1.0.10

2005-11-22 Thread Pedro
I noticed that asterisk.org now has asterisk and zaptel downloads for
version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9
version number. Just wondering for those using the 1.0.x versions
of asterisk instead of the 1.2 versions - will libpri, addons and
sounds be updated to match the 1.0.10 version or will 1.0.9 be the
final release of those packages?

- Pedro
___
--Bandwidth and Colocation sponsored by Easynews.com --

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: Asterisk 1.0.10

2005-11-22 Thread Pedro
Note - looks like the answer to this was posted out of *date* sequence on asterisk.org (it is below the 1.2.0 release notice):

 

 
direct from asterisk.org homepage:
Version
1.0.10 has been released of Asterisk and Zaptel. Libpri,
Asterisk-addons, and Asterisk-sounds contain no changes, so they have
not been updated.

It is very likely that this will be the final release of the 1.0
branch of Asterisk. Users are strongly encouraged to begin upgrading to
version 1.2.
Thanks!

 
  


On 11/22/05, Pedro [EMAIL PROTECTED] wrote:
I noticed that asterisk.org now has asterisk and zaptel downloads for
version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9
version number. Just wondering for those using the 1.0.x versions
of asterisk instead of the 1.2 versions - will libpri, addons and
sounds be updated to match the 1.0.10 version or will 1.0.9 be the
final release of those packages?

- Pedro


___
--Bandwidth and Colocation sponsored by Easynews.com --

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] How do you disable realtime?

2005-11-21 Thread Pedro
Am I correct in assuming that if I am not running Realtime on my
asterisk 1.2 server, the proper way to disable it is to remove the
following 2 files:

/usr/lib/asterisk/modules/pbx_realtime.so
/usr/lib/asterisk/modules/app_realtime.so

I am just testing out the default installation and am getting these errors on the console:

Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: Failed
to connect database server on . Check debug for more info.
Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug.

Any help will be appreciated.

- Pedro
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] How do you disable realtime?

2005-11-21 Thread Pedro
Yeah - tried that. Here are 2 lines I have in my modules.conf file:

noload = pbx_realtime.so
noload = app_realtime.so 
For some reason, I still get the following in my logs even after a restart of Asterisk.

Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime: Failed
to connect database server on . Check debug for more info.
Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug.

Any thoughts?

- Pedro
On 11/21/05, Alexander Lopez [EMAIL PROTECTED] wrote:





It is a better practice to use a noload option in 
modules.conf. That way if and when you upgrade you wont need to remove them 
again they will just continue to not load

Alex


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] On Behalf Of 
  PedroSent: Monday, November 21, 2005 12:11 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] How do you disable realtime?
  Am I correct in assuming that if I am not running Realtime on my 
  asterisk 1.2 server, the proper way to disable it is to remove the following 2 
  files:/usr/lib/asterisk/modules/pbx_realtime.so/usr/lib/asterisk/modules/app_realtime.soI 
  am just testing out the default installation and am getting these errors on 
  the console:Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL 
  RealTime: Failed to connect database server on . Check debug for more 
  info.Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: 
  Couldn't establish connection. Check debug.Any help will be 
  appreciated.- Pedro

___--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] How do you disable realtime?

2005-11-21 Thread Pedro
Thanks Bruce - but the whole point I am trying to accomplish is that I
don't want to use Realtime and don't want asterisk to try to establish
the connection. Was just chatting in IRC about this and it seems
that Realtime may not be able to be truly disabled (not sure how
accurate that is, but that was what I was told). Basically I just
want to have asterisk load without those 2 errors popping up on the
console and in the logs.On 11/21/05, Bruce Ferrell [EMAIL PROTECTED] wrote:
Check the mysql logs.I would suspect from this one of several things:1.) the userid/password is incorrect.on the db host use the command lin e mysql client like so: mysql -h localhost -u asterisk user -p
 you'll be prompted for a password.If that works, go to the next possible problem2.) the userid doesn't have correct permissions to the DBfrom the mysql client, issues the use command to try to access the
realtime DB.if that works, go to the next possible problem.3.) the userid is not permitted from the host the asterisk box is onas the mysql superuser look at mysql.user to see what hosts are
permitted access by the asterisk userid/password.If you have toadd a host, be sure to issue the flush priviledges commandPedro wrote: Yeah - tried that.Here are 2 lines I have in my 
modules.conf file: noload = pbx_realtime.so noload = app_realtime.so For some reason, I still get the following in my logs even after a restart of Asterisk.
 Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime: Failed to connect database serveron . Check debug for more info. Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime:
 Couldn't establish connection. Check debug. Any thoughts? - Pedro On 11/21/05, Alexander Lopez [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: It is a better practice to use a noload option in modules.conf. That way if and when you upgrade you wont need to remove them again they
 will just continue to not load Alex  From: 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Pedro Sent: Monday, November 21, 2005 12:11 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How do you disable realtime? Am I correct in assuming that if I am not running Realtime on my
 asterisk 1.2 server, the proper way to disable it is to remove the following 2 files: /usr/lib/asterisk/modules/pbx_realtime.so /usr/lib/asterisk/modules/app_realtime.so
 I am just testing out the default installation and am getting these errors on the console: Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime:

Failed to connect database serveron . Check debug for more
info. Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Any help will be appreciated. - Pedro
 ___ --Bandwidth and Colocation sponsored by Easynews.com 
http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:
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 
 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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___--Bandwidth and Colocation sponsored by 
Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

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] How do you disable realtime?

2005-11-21 Thread Pedro
Olle,
Yep - was actually replying to this as I got your message - I was
searching for modules that had realtime in the name (did not see the
res_config_mysql.so file). Setting the noload =
res_config_mysql.so in modules.conf took care of the issue I was having.

Thanks for your prompt response!

-PedroOn 11/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
Pedro wrote: Yeah - tried that.Here are 2 lines I have in my modules.conf file: noload = pbx_realtime.so noload = app_realtime.so For some reason, I still get the following in my logs even after a
 restart of Asterisk. Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime: Failed to connect database serveron . Check debug for more info. Nov 21 13:17:08 WARNING[31192] res_config_mysql.c: MySQL RealTime:
 Couldn't establish connection. Check debug. Any thoughts? - Pedro On 11/21/05, *Alexander Lopez* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote: It is a better practice to use a noload option in modules.conf. That way if and when you upgrade you wont need to remove them
 again they will just continue to not load Alex  *From:* 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Pedro *Sent:* Monday, November 21, 2005 12:11 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] How do you disable realtime? Am I correct in assuming that if I am not running Realtime on
 my asterisk 1.2 server, the proper way to disable it is to remove the following 2 files: /usr/lib/asterisk/modules/pbx_realtime.so /usr/lib/asterisk/modules/app_realtime.so
 I am just testing out the default installation and am getting these errors on the console: Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: Failed to connect database serveron . Check debug
 for more info. Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Any help will be appreciated.
Realtime is implemented in several places. PBX_realtime is the realtimeswitch, app_realtime is an application.res_config_mysql.c/so is the realtime driver for Mysql databases. So no,you are not correct. You have not removed all the
modules that involve realtime.On the other hand, the easiest way to disable realtime is not to enableit in the configuration file, extconfig.confYes, it's a strange name, but there are historical reasons for it :-)
/O___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Asterisk 1.2 Released!

2005-11-17 Thread Pedro
What jitterbuffer issues are you having with connecting to 1.0.x servers?On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED]
 wrote:Asterisk guy wrote: does it include the patch for VAD?
 ( dropping extra frame of G.729 since we already have a VAD frame at the end )It does not include several important things.It does not include a SIPjitter buffer.It does not include the ability to use Zaptel for timing
of the RTP audio.It does not include VAD/CND support.As far as Iknow it also does not have the patch to make the new IAX2 jitterbufferwork correctly when connecting to a 1.0.x server.___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

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] res_musiconhold.c: Music on Hold class 'default' already exists

2005-11-15 Thread Pedro
I just installed asterisk 1.2 rc2 and ran a 'make samples' and asterisk
starts just fine with no errors in the logs. However, if I issue
a reload I get the following:

Nov 15 17:08:22 WARNING[27009] res_musiconhold.c: Music on Hold class 'default' already exists

It is almost like the previous musiconhold process was not stopped
(guessing here as I am not a programmer)? Does this make sense?

Has anyone else seen this?
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Fritz!Card PCI ver2.0

2005-11-02 Thread Pedro Nunes

What chipset that card use??

Pedro Nunes


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: terça-feira, 1 de Novembro de 2005 23:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Fritz!Card PCI ver2.0

Anyone knows how I can use this ISDN card for asterisk as a BRI trunk
interface?


Thanks,
Stephen



___
--Bandwidth and Colocation sponsored by Easynews.com --

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
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Bristuff question

2005-10-27 Thread Pedro Nunes








Hi there,



I have 2 ISDN modems (HFC chipset). I use bristuff
from junghanns. Its
possible to load one cards as NT (T-Bus) and the other as S-Bus.



When I do make load the 2 cards loads
as S-Bus and when I do make loadNT the 2 cards loads as T-Bus.

Can someone help me??



Thanks in advance



Pedro Nunes






___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Bristuff question

2005-10-27 Thread Pedro Nunes








Giordano,



Thanks, stupid question.
Ive look to that page 100 of times but I do not remember that part of
the page about loading more than one card :S.



Thanks again









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Giordano Grandis
Sent: quinta-feira, 27 de Outubro
de 2005 15:30
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: R: [Asterisk-Users]
Bristuff question





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



look this





Giordano 











Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes
Inviato: giovedì 27 ottobre 2005
16.23
A: asterisk-users@lists.digium.com
Oggetto: [Asterisk-Users] Bristuff
question





Hi there,



I have 2 ISDN modems (HFC chipset). I use bristuff
from junghanns. Its
possible to load one cards as NT (T-Bus) and the other as S-Bus.



When I do make load the 2 cards loads
as S-Bus and when I do make loadNT the 2 cards loads as T-Bus.

Can someone help me??



Thanks in advance



Pedro Nunes






___
--Bandwidth and Colocation sponsored by Easynews.com --

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] ACD/queues question

2005-10-13 Thread Pedro Nunes
Thanks,

That will fix my problem... And agent skills, is that possible too??

Thanks again

Pedro Nunes

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lorenzo
Emilitri
Sent: quinta-feira, 13 de Outubro de 2005 8:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ACD/queues question


Hello Pedro,
you should do this using agent priority groups; this way first all low  
priority agents are filled, then another group is used up.
Thanks
l.


On Wed, 12 Oct 2005 19:30:43 +0200, Pedro Nunes [EMAIL PROTECTED]

wrote:

 Hi there,


 Does anyone know how to setup an overflow queue? When a call rings on
 the queue A, if all agents were busy, the call goes to the queue B.

 If all agents in queue B were busy, then the call stays on both queues
 until somebody answers it.


 I think this is a basic ACD feature available on most PBX that support
 ACD functionality.

 Does anybody knows how to do it with asterisk??



 Thanks in advance



 Pedro Nunes





-- 
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
___
--Bandwidth and Colocation sponsored by Easynews.com --

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
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] ACD/queues question

2005-10-13 Thread Pedro Nunes
Thanks,

That will fix my problem... And agent skills, is that possible too??

Thanks again

Pedro Nunes

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: quarta-feira, 12 de Outubro de 2005 23:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ACD/queues question

On Oct 12, 2005, at 1:30 PM, Pedro Nunes wrote:
 Hi there,

 Does anyone know how to setup an overflow queue? When a call rings  
 on the queue A, if all agents were busy, the call goes to the queue B.

 If all agents in queue B were busy, then the call stays on both  
 queues until somebody answers it.

 I think this is a basic ACD feature available on most PBX that  
 support ACD functionality.

 Does anybody knows how to do it with asterisk??

 Thanks in advance

  Pedro Nunes
What we have done is to set up a single queue that all calls come  
into. For the agents that we want to be our Front Line (i.e.:  
Customer Service Reps), we give them a penalty of 0. Our Overflow  
group (i.e.: Customer service reps who are also dealing with walk-in  
customers and therefore should not be bothered unless we're really  
busy) gets a penalty of 1, and our Last Resort (i.e.: Everyone  
else) people get a penalty of 2.

That way, all of the calls are answered by our front line people,  
unless they are all busy/unavailable. Then, and only then, the calls  
start going to our overflow people, and if they are also all  
unavailable, the calls go to our last resort people. Seeing as how we  
have more than 23 people between the three groups, there should  
technically be no waiting on hold in the queue, even with the PRI  
saturated.

I don't know if this is what you are looking for, but it works  
extremely well for us. To whomever coded this feature, THANK YOU!

To set this up, just edit the queues.conf file and add the penalty to  
each agent's  member = line like this:

; Front-line - Penalty of 0
member = 100,0
; Overflow - Penalty of 1
member = 101,1
;Last Resort - Penalty of 2
member = 102,2

Hope that proves useful to someone

Tom

___
--Bandwidth and Colocation sponsored by Easynews.com --

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
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] AGI and set_callerid for number and name

2005-10-12 Thread Pedro Nunes
Curse,

Look at this php script ...

Contactlookup.agi

#!/usr/local/bin/php -q
 ?php
 ob_implicit_flush(true);
 set_time_limit(6);
 $in = fopen(php://stdin,r);
 $stdlog = fopen(/var/log/asterisk/my_agi.log, w);

 // toggle debugging output (more verbose)
 $debug = true;

 // Do function definitions before we start the main loop
 function read() {
   global $in, $debug, $stdlog;
   $input = str_replace(\n, , fgets($in, 4096));
   if ($debug) fputs($stdlog, read: $input\n);
   return $input;
 }

 function errlog($line) {
   global $err;
   echo VERBOSE \$line\\n;
 }

 function write($line) {
   global $debug, $stdlog;
   if ($debug) fputs($stdlog, write: $line\n);
   echo $line.\n;
 }

 // parse agi headers into array
 while ($env=read()) {
   $s = split(: ,$env);
   $agi[str_replace(agi_,,$s[0])] = trim($s[1]);
   if (($env == ) || ($env == \n)) {
 break;
   }
 }

 // main program
 echo VERBOSE \Here we go!\ 2\n;
 read();
 $session = mssql_connect('mssql server' , 'username' , 'password' );
$result = mssql_query(select * from ContactDB WHERE
extension=.$agi['callerid'],$session );
 $row = mssql_fetch_array($result);
 mssql_close($session);
 if ($row['Name'] == ){
  write('SET VARIABLE NAME Not Found');
  read();
 } else {
  write('SET VARIABLE NAME '.$row['Name'].'');
  read();
 }
 fclose($in);
 fclose($stdlog);


And in extensions.conf

[extensions]
exten = 4501,1,agi,contactlookup.agi
exten = 4501,2,SetCIDName(${Name})
exten = 4501,3,Dial(SIP/421,15)


It looks to an mssql DB, try to find the callerID number in table
extensions, and then sets a variable named Name to the value of
table Name. Cool hah...



Pedro Nunes



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Lhermitte
Sent: quarta-feira, 12 de Outubro de 2005 17:57
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AGI and set_callerid for number and name


Hi,


I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber. 

Is it at at possible ?

Cheers.
SL


 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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
___
--Bandwidth and Colocation sponsored by Easynews.com --

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] ACD/queues question

2005-10-12 Thread Pedro Nunes










Hi there,



Does anyone know how to
setup an overflow queue? When a call rings on the queue A, if all agents were
busy, the call goes to the queue B.

If all agents in queue B were
busy, then the call stays on both queues until somebody answers it. 



I think this is a basic
ACD feature available on most PBX that support ACD functionality. 

Does anybody knows how to
do it with asterisk??





Thanks in advance





Pedro Nunes










___
--Bandwidth and Colocation sponsored by Easynews.com --

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] WiFi Phones

2005-10-10 Thread Pedro
The UTStarcom F1000 with the latest firmware (3.10st) has improved
sound volume over the default firmware shipped with the units.
Also, TFTP configuration works well so you don't have to configure the
units with the keypad. You will need to get the configuration
compiler from your vendor and be aware that the default encryption key
should be set to NULL rather than F1000 as stated in the docs when
compiling your config. At first I was not sure how I would like a
WiFi phone because I figured it would sound bad, but I have been very
impressed with the quality of the F1000. We have now added it to
our VoIP product offerings.

- Pedro
http://www.traci.netOn 10/8/05, Cory Andrews [EMAIL PROTECTED] wrote:
The F3000 is not anticipated to be available for distribution until lateDecember/January, FYI.
Cory AndrewsSenior Partner+++VOIPSupply.com454 Sonwil DriveBuffalo, NY 14225+++voice - 716.630.1555 X22email - [EMAIL PROTECTED]
fax - 716.630.1548Denis Galvão - iSolve wrote: Wait for the next UTStarCom version... Called F3000, Im not sure, but something like that. It will have better battery performance and will have 
802.11g support, and many other improvements. It will be available soon. Denis. On 07 de out de 2005, at 00:54, Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsetscurrently
 available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP
 option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the
 language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone
 is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my
 fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 
802.1x authentication, it sure looks like WEP is the only available security option. Overall: I would recommend purchasing one, for testing at the very least.
They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by 
Easynews.com -- 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
 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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___
--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

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] Logging SIP response codes

2005-07-19 Thread Pedro
Had not seen a response on the following question - wondering if
anyone may have any insight on this?

Original Question-
Is there a way to log SIP response codes without enabling verbose
logging?  Reason being is that from time to time I see a call fail on
our primary provider and roll-over to our backup providers.  If I
happen to catch it on the console I can see the code 484 or similar.
 It would really help in troubleshooting with our primary provider if
I could log those types of codes.  Verbose just saves way to much
stuff in the log files.
___
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] Logging SIP response codes

2005-07-07 Thread Pedro
Is there a way to log SIP response codes without enabling verbose
logging?  Reason being is that from time to time I see a call fail on
our primary provider and roll-over to our backup providers.  If I
happen to catch it on the console I can see the code 484 or similar.
 It would really help in troubleshooting with our primary provider if
I could log those types of codes.  Verbose just saves way to much
stuff in the log files.
___
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] how to send voicemail notifcation every 15 minutes until message is checked

2005-07-01 Thread Pedro
I have searched quite a few places and have not seen this discussed. 
Basically I was wondering how would you go about having an option for
a user to be notified every 15 minutes until their new voicemail
message is checked.  Since the notification e-mails we send get sent
to cell phones or actual pagers (via e-mail), there are times when a
person is out of range and misses a page or just simply is too busy to
check voicemail and then forgets.  They want to be reminded 15 minutes
later until that new message is checked.

Current version of asterisk that we are running is CVS-v1-0-11/12/04
(which has been running rock-solid I might add).  Any thoughts are
appreciated.
___
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] how to send voicemail notifcation every 15 minutes until message is checked

2005-07-01 Thread Pedro
Thanks - a cronjob for the user was going to be my last resort.  Was
not sure if there was a setting like repeatnotify=15 to repeat the
notice every 15 minutes.

Thanks for your feedback though!

On 7/1/05, Michiel van Baak [EMAIL PROTECTED] wrote:
 On 13:33, Fri 01 Jul 05, Pedro wrote:
  I have searched quite a few places and have not seen this discussed.
  Basically I was wondering how would you go about having an option for
  a user to be notified every 15 minutes until their new voicemail
  message is checked.  Since the notification e-mails we send get sent
  to cell phones or actual pagers (via e-mail), there are times when a
  person is out of range and misses a page or just simply is too busy to
  check voicemail and then forgets.  They want to be reminded 15 minutes
  later until that new message is checked.
 
  Current version of asterisk that we are running is CVS-v1-0-11/12/04
  (which has been running rock-solid I might add).  Any thoughts are
  appreciated.
 
 Hi,
 
 You can check the new mail count with the manager interface
 or by looking at the spool dir.
 If you put this in cron every 15 minutes, you're done.
 
 Michiel
 ___
 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] Anyone noticed Voipjet voice quality problems?

2005-06-29 Thread Pedro
Looks like 9 out of 10 calls are failing on voipjet at the moment (at
least terminating to South Florida numbers).  Keep getting message
that says number can not be completed as dialed.  Anyone else seeing
this?

On 6/15/05, Pedro [EMAIL PROTECTED] wrote:
 Couple of days.  Apparently the new US carrier has some changes that
 needs to be made.
 
 On 6/14/05, Wiley Siler [EMAIL PROTECTED] wrote:
  Did they say when it would be corrected?
 
  W
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Pedro
  Sent: Tuesday, June 14, 2005 9:22 AM
  To: Matt
  Cc: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality
  problems?
 
  Caller ID is still not working to certain areas.  This problem was
  confirmed by voipjet tech support in their last e-mail to me.
 
  On 6/13/05, Matt [EMAIL PROTECTED] wrote:
   I never noticed any problems.. so I can't comment :) hehe
  
   On 6/11/05, Pedro [EMAIL PROTECTED] wrote:
Finally got a response from voipjet support and they say they have
switched to a new provider for US termination.  I have yet to test
this out as I have not had a chance to build them back into our
routes but will report my findings once I do.  Anyone else notice
any improvements?
   
On 6/9/05, Moody [EMAIL PROTECTED] wrote:
 We have been having serious quality problems using the westcoast
 server - been using the East coast server with increased success
 but seeing some issues related to going cross continent.

 Voipjet, you listening?
 ___
 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 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] SIP connection

2005-06-16 Thread Pedro Diaz



I need help to make a conection form FWD to my pbx, 
I can receive a call from PSTN for a FXo card but know I need to receive call 
via IP form FWD I have activate hte IAX on freeworlddialup but does not work I 
can't make or receive calls. I virtually new in this can please somebody help 
me.

thanks,

scorpionny
___
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] Anyone noticed Voipjet voice quality problems?

2005-06-15 Thread Pedro
Couple of days.  Apparently the new US carrier has some changes that
needs to be made.

On 6/14/05, Wiley Siler [EMAIL PROTECTED] wrote:
 Did they say when it would be corrected?
 
 W
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Pedro
 Sent: Tuesday, June 14, 2005 9:22 AM
 To: Matt
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality
 problems?
 
 Caller ID is still not working to certain areas.  This problem was
 confirmed by voipjet tech support in their last e-mail to me.
 
 On 6/13/05, Matt [EMAIL PROTECTED] wrote:
  I never noticed any problems.. so I can't comment :) hehe
 
  On 6/11/05, Pedro [EMAIL PROTECTED] wrote:
   Finally got a response from voipjet support and they say they have
   switched to a new provider for US termination.  I have yet to test
   this out as I have not had a chance to build them back into our
   routes but will report my findings once I do.  Anyone else notice
   any improvements?
  
   On 6/9/05, Moody [EMAIL PROTECTED] wrote:
We have been having serious quality problems using the westcoast
server - been using the East coast server with increased success
but seeing some issues related to going cross continent.
   
Voipjet, you listening?
___
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 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] Anyone noticed Voipjet voice quality problems?

2005-06-14 Thread Pedro
Caller ID is still not working to certain areas.  This problem was
confirmed by voipjet tech support in their last e-mail to me.

On 6/13/05, Matt [EMAIL PROTECTED] wrote:
 I never noticed any problems.. so I can't comment :) hehe
 
 On 6/11/05, Pedro [EMAIL PROTECTED] wrote:
  Finally got a response from voipjet support and they say they have
  switched to a new provider for US termination.  I have yet to test
  this out as I have not had a chance to build them back into our routes
  but will report my findings once I do.  Anyone else notice any
  improvements?
 
  On 6/9/05, Moody [EMAIL PROTECTED] wrote:
   We have been having serious quality problems using the westcoast
   server - been using the East coast server with increased success but
   seeing some issues related to going cross continent.
  
   Voipjet, you listening?
   ___
   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 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] Anyone noticed Voipjet voice quality problems?

2005-06-11 Thread Pedro
Finally got a response from voipjet support and they say they have
switched to a new provider for US termination.  I have yet to test
this out as I have not had a chance to build them back into our routes
but will report my findings once I do.  Anyone else notice any
improvements?

On 6/9/05, Moody [EMAIL PROTECTED] wrote:
 We have been having serious quality problems using the westcoast
 server - been using the East coast server with increased success but
 seeing some issues related to going cross continent.
 
 Voipjet, you listening?
 ___
 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] Anyone noticed Voipjet voice quality problems?

2005-06-10 Thread Pedro
Seems things have just got worse.  Just got reports that 800 numbers
are not terminating.  For example, can not dial:

800-888-9358
or
800-922-4684

Had to pull voipjet out of our routes until this gets fixed.

On 6/9/05, Moody [EMAIL PROTECTED] wrote:
 We have been having serious quality problems using the westcoast
 server - been using the East coast server with increased success but
 seeing some issues related to going cross continent.
 
 Voipjet, you listening?
 ___
 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] Anyone noticed Voipjet voice quality problems?

2005-06-10 Thread Pedro
We are a VoIP provider and need to push out 100,000  - 200,000 minutes
per month (ie. need a carrier-level package - not a Vonage, etc.).  To
date I have not found a wholesale SIP/IAX VoIP provider provide 800
termination for free.  However, if you have one, please provide the
information and I will definately check them out.

On 6/10/05, Pedro [EMAIL PROTECTED] wrote:
 Please provide the SIP or IAX provider you are using that allows you
 to terminate to 800 numbers for free.
 
 On 6/10/05, Matt [EMAIL PROTECTED] wrote:
  Why would you even be routing 800 numbers out voipjet?  They CHARGE you!
 
  On 6/10/05, Pedro [EMAIL PROTECTED] wrote:
   Seems things have just got worse.  Just got reports that 800 numbers
   are not terminating.  For example, can not dial:
  
   800-888-9358
   or
   800-922-4684
  
   Had to pull voipjet out of our routes until this gets fixed.
  
   On 6/9/05, Moody [EMAIL PROTECTED] wrote:
We have been having serious quality problems using the westcoast
server - been using the East coast server with increased success but
seeing some issues related to going cross continent.
   
Voipjet, you listening?
___
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 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] Anyone noticed Voipjet voice quality problems?

2005-06-09 Thread Pedro
Definately problems with voice quality and caller ID is not working
very well.  I have e-mail a couple times and still no response from
their tech support on this.  This is very concerning since I tried all
3 servers with the same results.

On 6/8/05, Julio Arruda [EMAIL PROTECTED] wrote:
 Roman Zhovtulya wrote:
  Dear all,
  I've noticed some significant voice quality deterioration when calling US
  landline via VoIPjet.com in the last week or so.
  Before that the quality was pretty good.
  Has anyone else experienced any voice quality problems with voipjet
  recently?
 
 I've been using VOIPJET for Brazil LD without any problems.
 (or should I say, my wife has been using, still can't thank VOIP enough
 for the savings..)
 ___
 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] How to detect DTMF and change if needed

2005-05-23 Thread Pedro
I have done some searching and not sure this is even possible, but
here it goes...

**Scenario**
Let's say you have an asterisk server that you use to connect to a SIP
provider that you push your PSTN-bound calls to using g711 and
out-of-band DTMF.  The SIP phones in use are Cisco 7960's and are set
to also use out-of-band DTMF.  For the most part, everything works
great.

However, a few numbers that are dialed and pushed to the SIP provider
that get connected to a remote IVR system seem to have DTMF issues
where no digits are recognized.  A call to the SIP provider confirms
that certain calls get routed to one carrier while others get routed
to other carriers and the numbers that are showing the DTMF issues are
the carriers that they peer with that do not support out-of-band DTMF
with the g711 codec.  When asked if they could translate our
out-of-band DTMF signals to a compatible format that their carrier
requires, they bascally say that while that is possible, they will not
do it.

**The Question**
So here is my question - is it possible to detect the DTMF mode of the
call and if out-of-band is not supported, can you change it to inband
as a last resort?

Is there a way to set priority for DTMF signalling like you can do
with codecs?  I have tried that (see below) but it seems to default to
inband (is this even a proper way to handle 2 DTMF modes?).

[sipprovider]
type=friend
host=xxx.xxx.xxx.xxx
disallow=all  
allow=ulaw
maxexpirey=15
dtmfmode=rfc2833
dtmfmode=inband
nat=no
insecure=very
canreinvite=no

I have searched and searched and the closest thing that I have found
is SIPDtmfMode but from what it looks like it needs to be initiated
before the call is placed.

By the way - the reason inband is not being used is that digit
accuracy is terrible with the inband setting.

Any thoughts are appreciated.
___
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] FCC Will Force VOIP E911 in 120 days ?

2005-05-18 Thread Pedro
http://www.lightreading.com/document.asp?doc_id=73943site=lightreading

I know e911 has been discussed on ths list before, but I just read
this and it got me thinking that if you have a Wholesale VoIP carrier
- wouldn't they have to pass e911 on to you as a VoIP provider to, in
turn, pass on to your end-users?  Of course there would be a fee -
just wondering if this is how the start-ups will be able to reach a
deadline on this if it passes.  Especially since it seems you have to
be a CLEC to interface with the PASP database from the threads I have
been reading.

Also, how in the world will this work with hosted IP-PBX solutions
where the customer may have their employees scattered around the
country working from their homes?  Since all their outbound calls
share the same callerID which may not even be in the local area that
they are physically located in, how will the call get routed to the
proper PASP (and better yet - how will the PASP know which employee
called them)?
___
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] g729 license

2005-05-02 Thread Pedro
Actually called Digium with this exact question last week.  They said
that you can register the new license on the new server provided that
you ony registered it once before.  They said there is no unregister
script to unregister the license from the old server, however.  If you
have already used up your 2 registrations, you will need to contact
Digium for assistance on this.  I also asked if leaving the keys on my
dev. box would cause a conflict (also was pretty clear that I wanted
to be in compliance with their license agreement) and the lady said
there was no problem and leaving the old keys on the dev. box would
not cause a conflict.

On 5/2/05, Peter [EMAIL PROTECTED] wrote:
 Hi all.
 
 Dopes someone know how I can move a key license of the g729
 codec from one to another machine?
 Find nothing usefull @ the wiki.
 
 Thnx 4 help in advance.
 
 Regards.
 
 -Peter
 
 --
 Please no HTML, I'm not a browser
 
 ___
 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] A good SIP receptionist phone

2005-05-02 Thread Pedro
What I did once was create an announcement that got played to the
receptionist announcing who the call was for based on the number that
was called.  This allowed the receptionist to know which greeting to
recite.

On 5/2/05, Michael Welter [EMAIL PROTECTED] wrote:
 Chris Mason (Lists) wrote:
  The user name is the extension and the password is always the same. Not hard
  to configure.
 
 With the SNOM 220, you have five buttons/lamps that can be used as
 line appearances--these buttons can each register to a different SIP URL.
 
 Each sidecar has 20 buttons/lamps, and you may have up to three
 sidecars.  Using the hint priority in Asterisk, the buttons serve as
 extension busy lamps.  You can also use these buttons to transfer calls.
 
 I have an executive suites customer where each tenant is a separate
 business.  For an incoming call, the attendant needs to know which DID
 number is being called so she can answer with the proper greeting.
 
 I would like the sidecar buttons to be able to register to a SIP URL, so
 an incoming call would blink the tenants button, but that is not
 possible--I can only use the five buttons on the phone for that purpose,
 and there are more than five tenants.
 
 A suggestion was to alter the Called ID Name to the DID number.  This
 would work for the attendant, but the tenant would like to see the
 original Caller ID Name.
 
 I would rather not have to put a PC at the attendants position, but that
 is the way this is shaping up.  Does anyone have any suggestions?
 
 Thanks,
 
 
 ___
 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] Confused on G723 and G729

2005-04-28 Thread Pedro
In your case, where you will need the license is on the box that your
phones register to.  For exampe, when someone checks voicemail,
encoding takes place, therefore you need a license.

Look at it this way:

[g729 provider] -(SIP or IAX)--- [g729 asterisk server]

- no license required in the above connection if using g729 solely

[g729 asterisk server containing non-g729 audio files]
(SIP)- [g729 SIP Phone]

- a license is required above for each non-g729 audio file or stream
that needs to be encoded to be sent out as g729 to the g729 SIP Phone
(ie. voicemail, IVR prompts, etc.).

Hope that makes sense.




On 4/28/05, Matt [EMAIL PROTECTED] wrote:
 For instance.. when I try to use G723.1 on my phone (and just call in
 from my PRI line) I get:
 Unable to find a path from g723 to ulaw.
 Unable to find a path from ulaw to g723.
 No path to translate from Zap/1-1(68) to Sip/201-80c7(1).
 Same things happens if I call in on my current provider's number which
 uses G711 for the codec.
 ___
 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] Mitel Ip phone ?

2005-04-26 Thread Pedro
Just wanted to correct this last post - apparently, you can configure
the other speed buttons to also be separate lines with their own SIP
account.

On 4/22/05, Pedro [EMAIL PROTECTED] wrote:
 Just got the new Mitel 5220 Dual Mode IP Phone and I have to say I am
 impressed so far.  Changing to SIP mode is VERY easy as long as you
 have the SIP firmware which can be downloaded by going to:
 
 http://sipdnld.mitel.com
 
 Phone audio quality is excellent.  Look and feel are also good.  The
 phone can only accept 1 SIP user account, but can handle 4
 simultaneous calls.  Conferencing can be done with 2 of the 4
 simultaneous calls and you can switch between any of the calls at any
 time.  The only issue I see is the small display.  Clearing a missed
 call involves cycling through a few menus to clear the missed call log
 (not sure there is a short cut for this).  The speed dial buttons
 fine, but you must manually write the name of each speed dial button
 (Cisco-type LCD would be nice for this but would probably add to the
 cost of the phone).
 
 Well, that is my initial impressions.  If I come across anything else
 important I will let you know.
 
 Pedro
 TRACI.net
 
 On 4/4/05, Kris Edwards [EMAIL PROTECTED] wrote:
  Here's a good sign:
 
  Mitel is also addressing economy in adding SIP compliance to two of its
  IP phones. The Mitel 5215 and 5220 run Mitel's proprietary MiNET
  protocol for operation with the ICP 330 but also will run SIP, allowing
  users to point them at such SIP-based PBXes as Asterisk's or Snom's, or
  to another SIP proxy server.
 
  _
 
  I know Mitel's older models can be changed to sip (there's a howto on
  voip-info) so surely there is hope for the newer models.
 
  Here's the article the above is taken from:
 
  http://www.thechannelinsider.com/article2/0,1759,1725518,00.asp
 
 
  Kris
 
 
  Pedro wrote:
   Will let you know - getting one soon to test.
  
   On Feb 13, 2005 10:58 PM, eric m [EMAIL PROTECTED] wrote:
  
  Hi,
  
  I really appreciate the look and design of newer Mitel Ip phone.
  
  I search througt the list and found only fews notes about the use Mitel 
  5055
  phone on *.   Anyone use other model (especially 52xx series) on * ??
  Compatible?  Easy to use?  hassle to configure?
  
  Thansk for your suggestion!
  
  Best Regards,
  
  eric.
  [EMAIL PROTECTED]
  
  ___
  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 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] DTMF intermittently stops working

2005-04-25 Thread Pedro
Thank you for your feedback.

I was mearly wondering if others had experienced this issue in their
environments.  Was not trying to open a bug report or officially
report an issue.  Strictly a curiousity request.  Really do not want
to upgrade if everything else works fine.  Since this issue happens so
intermittently, I would have no way of testing if the new version
would fix it since I could go for 6 months without having the issue on
my current version (no way to consistently replicate the problem).  If
you have a way to consistently replicate this issue, I would
appreciate that information.

I can assure you I exhausted search options and researched this issue
elsewhere with little success before posting my question here to avoid
wasting people's time.

On 4/24/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
 Joseph wrote:
 
  We have the same problem with 7960, just randomly it will stop *hearing*
  the dtmf tones and you have to hangup and call back.
 
 This problem was fixed in CVS long ago, and current stable releases have
 the fix as well. When you are running a copy of Asterisk that is 4/5
 months old, it's better to update first before reporting a problem,
 since it may already have been fixed.
 ___
 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] SIP registration behind Linksys WRT54G

2005-04-23 Thread Pedro
Have you tried to enable NAT translation on the Grandstream?

On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote:
 I'm trying to register BT100s ... (doesn't work)
 X-Lite seems to work though
 
 Tomas
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo
 Sent: Saturday, April 23, 2005 8:48 PM
 To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
 
 Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running
 behind my Linksys WTR43GS with no issues. This is at home registering to an
 external * box and to vonage.
 
 - Original Message -
 From: Luki [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Saturday, April 23, 2005 9:41 PM
 Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G
 
 The WRT54G work fine...
 
 I have a Sipura 1000 and a Grandstream 286, both nated through a
 WRT54G on a single public IP. Worked out of the box -- no special
 settings needed. I was even surprised that I did not need to turn on
 the NAT handling in the Sipura ATA.
 
 Then I have a WRT54G running as a wireless client, and a Sipura 1001
 connected to it, essentially behind two NAT's. Works fine too.
 
 --Luki
 ___
 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 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] Mitel Ip phone ?

2005-04-22 Thread Pedro
Just got the new Mitel 5220 Dual Mode IP Phone and I have to say I am
impressed so far.  Changing to SIP mode is VERY easy as long as you
have the SIP firmware which can be downloaded by going to:

http://sipdnld.mitel.com

Phone audio quality is excellent.  Look and feel are also good.  The
phone can only accept 1 SIP user account, but can handle 4
simultaneous calls.  Conferencing can be done with 2 of the 4
simultaneous calls and you can switch between any of the calls at any
time.  The only issue I see is the small display.  Clearing a missed
call involves cycling through a few menus to clear the missed call log
(not sure there is a short cut for this).  The speed dial buttons
fine, but you must manually write the name of each speed dial button
(Cisco-type LCD would be nice for this but would probably add to the
cost of the phone).

Well, that is my initial impressions.  If I come across anything else
important I will let you know.

Pedro
TRACI.net

On 4/4/05, Kris Edwards [EMAIL PROTECTED] wrote:
 Here's a good sign:
 
 Mitel is also addressing economy in adding SIP compliance to two of its
 IP phones. The Mitel 5215 and 5220 run Mitel's proprietary MiNET
 protocol for operation with the ICP 330 but also will run SIP, allowing
 users to point them at such SIP-based PBXes as Asterisk's or Snom's, or
 to another SIP proxy server.
 
 _
 
 I know Mitel's older models can be changed to sip (there's a howto on
 voip-info) so surely there is hope for the newer models.
 
 Here's the article the above is taken from:
 
 http://www.thechannelinsider.com/article2/0,1759,1725518,00.asp
 
 
 Kris
 
 
 Pedro wrote:
  Will let you know - getting one soon to test.
 
  On Feb 13, 2005 10:58 PM, eric m [EMAIL PROTECTED] wrote:
 
 Hi,
 
 I really appreciate the look and design of newer Mitel Ip phone.
 
 I search througt the list and found only fews notes about the use Mitel 5055
 phone on *.   Anyone use other model (especially 52xx series) on * ??
 Compatible?  Easy to use?  hassle to configure?
 
 Thansk for your suggestion!
 
 Best Regards,
 
 eric.
 [EMAIL PROTECTED]
 
 ___
 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 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] NuFone problems to non-na numbers

2005-04-20 Thread Pedro
Yes, same problem here.  Sign-ed up with VoipJet and seems to work
just fine (prices for most areas we call are cheaper too from what I
saw).  Only been using them for 24 hours so can't say much about
long-term stability, but so far so good.

Pedro

On 4/19/05, Matthew Asham [EMAIL PROTECTED] wrote:
 Is anyone else having problems with Nufone dialing international (non
 NA) numbers?
 
 Pretty much every intl number dialed comes up with a voice intercept
 saying the call could not be completed as dialed.  Tried it with two
 separate accounts, and the numbers themselves work from the local
 telco.
 
 The problem appears to have started within the last few days (and yes I
 have emailed [EMAIL PROTECTED], just wondering if we're the only ones
 having the problem).
 
 Matthew
 
 --
 Matthew Asham - the B.C. Wireless Network Society
 www.bcwireless.net - +1 604 484 5289 x1006
 
 ___
 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] DTMF intermittently stops working

2005-04-18 Thread Pedro
Every so often I get a report from a customer that DTMF stops working
while checking voicemail.  The customer has to hang up and check for
messages again.  I have actually had this happen to me twice in the
past 6 months so I know it does happen, just not very often.

So far, the only incidents have been with Cisco 7960's.  I was just
wondering if anyone had noticed this behavior in their environment. 
We are using ulaw and rfc2833 with the following configuration
(Asterisk CVS-v1-0-11/12/04):

SIP Provider (SIP)(SIP) Asterisk Gateway (IAX)(IAX) Customer
Asterisk Server (SIP)(SIP) Cisco 7960


Any thoughts are appreciated.

Thank You,
Pedro
TRACI.net
___
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] Mitel Ip phone ?

2005-04-01 Thread Pedro
Will let you know - getting one soon to test.

On Feb 13, 2005 10:58 PM, eric m [EMAIL PROTECTED] wrote:
 Hi,
 
 I really appreciate the look and design of newer Mitel Ip phone.
 
 I search througt the list and found only fews notes about the use Mitel 5055
 phone on *.   Anyone use other model (especially 52xx series) on * ??
 Compatible?  Easy to use?  hassle to configure?
 
 Thansk for your suggestion!
 
 Best Regards,
 
 eric.
 [EMAIL PROTECTED]
 
 ___
 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] Cisco 7960 SIP Firmware

2005-03-18 Thread Pedro
Just a note that you will need to perform quite a few incremental
upgrades to get to a current firmware version.  So if you do get
someone who will sell you the firmware, make sure you get the all of
them.

On Fri, 18 Mar 2005 22:04:29 -0500, Patrick M. Gray, Jr.
[EMAIL PROTECTED] wrote:
  
  
 
 I got a new old stock Cisco 7960 from eBay and the warranty expired bay in
 2001 according to Cisco (I didn't even know they had VoIP in 2001 ;-) ).  I
 spoke with a wonderfully rude gentleman at Cisco who told me there was
 nothing that could be done to get SIP firmware for the device, and would not
 even entertain the possibility of purchasing said FW from Cisco.  He
 suggested I call a local reseller, and the single one I called was not
 interested in helping me either with my unsupported hardware. 
 
   
 
 I'm using the 7960 to experiment with *, and was wondering if there are
 alternative means to finding the firmware, or if the out of the box SCCP
 firmware (I have version P003AM30) will work with *.  I'm willing to pay any
 official resellers a fair price for the F/W, but the attitude I received
 from Cisco and the one reseller I contacted have me thinking this is a waste
 of time. 
 
   
 
 I'm using [EMAIL PROTECTED] and can seem to find any SCCP info, and don't want
 to delve too deeply into this experiment if the phone is not going to work
 reliably. 
 
   
 
 Thanks for any help or pointers in the right direction. 
 
   
 
 Pat 
 ___
 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] SIP VoIP Provider problems

2005-03-05 Thread Pedro
Sounds like you are having a codec issue with 2 of  your providers. 
Make sure you find out what codecs are supported and that your config
is set up accordingly.


On Sun, 06 Mar 2005 00:14:05 +, w fm3 [EMAIL PROTECTED] wrote:
 Hi
 
 Hope someone can help :)
 
 I am testing 4 PSTN termination providers. 3 SIP and 1 IAX
 
 IAX and 1 of the SIP providers work fine.
 
 Now the wierdness:
 
 2 SIP providers I can only get oubound calls to ring at the destination and
 then nothing more. 1 gets as far as SIP code 183 (and ringing on the src
 handset ...yay) the other doesn't get past 100.
 
 Added to this inbound calls (PSTN-provider-asterisk-handset) work fine
 100% of the time.
 
 I have tried alot of config options from the wiki and lists but can't seem
 to get any further.  AFAIK from sip debug and the console it looks like
 that the call is placed  and then no further  communication. Looks like they
 might be using SER / CISCO GW at the VOIP Provider end.
 Don't think it a open UDP port type thing.
 
 Cheers
 
 Walt
 
 PS Newbie
 
 _
 Express yourself instantly with MSN Messenger! Download today it's FREE!
 http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
 
 ___
 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] Zombie SIP channels

2005-03-04 Thread Pedro
Ok - I finally found out what was causing the ZOMBIE channels.

Now follow me on this one :)

It appears that if you are using a Cisco 7960 and are on a call and
want to transfer the call to another extension - if you press more
and Trnsfer and dial the extension and you hit the Trnsfer button
again before the extension answers, a ZOMBIE channel is created.

If you use BlindXfer, it does not create the ZOMBIE channel.

I have now informed my client that if they want to do a Blind
Transfer, to use the BlindXfer softkey instead of the Trnsfer softkey
or just use the # key to do a blind transfer.

Now, I am running Asterisk CVS-v1-0-11/12/04-15:32:45. I would be
interested in knowing if later versions of asterisk exhibited this
same behavior.  Any feedback would be appreciated.

Thanks,
Pedro


On Fri, 11 Feb 2005 08:32:43 +0100, Florian Overkamp
[EMAIL PROTECTED] wrote:
 Hi,
 
  -Original Message-
  Ok this is odd - caught it again twice today.  The more I thought
  about what has changed on the server I realized that I was not using a
  timing device before, but am now using ztdummy.  I if that could be
  causing the zombies?
 
http://bugs.digium.com/bug_view_page.php?bug_id=0002938
 
 I don't think so, but who knows. The patch resolves a locking issue that may
 or may not be timing-source dependant. I've seen the issue occur after call
 transfers in scenario's where I used a few chan_local's.
 
 Do yourself a favour:
 
 - If you can, unload the ztdummy and test for a while. However, this may put
 the issue to sleep - but it won't solve it!
 - After that, load ztdummy again and apply the two lines in channel.c. Test
 again. Good chance the issue will be gone.
 
 Report results here :)
 
 Florian
 

___
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] Asterisk SKINNY with Cisco IP Conference 7935

2005-03-02 Thread Pedro Mansilla








Hi,



 I have one Cisco IP
Conference 7935. Im trying to config
the SKINNY Protocol.



 I config
the skinny.conf file same like sample for Cisco 7910.



 When
somebody call me my phone ring and answer the call but I cant hear
anything,

 But the
other people is hearing me very good.



 When I try
to call somebody the * show me an error : RECEIVED UNKNOWN MESSAGE TYPE:
4



 Somebody have
a skinny.conf sample file for Cisco 7935 or any trick to fix this
problem



Thanks,



Pedro.








___
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] T.38 fax summary

2005-02-28 Thread Pedro Miguel de Sousa Caria
Hi, my name is Pedro Caria I'm new to this list.
I live in Portugal and find myself in the position to talk often to 
various parts of the world, very often the Telco line has a delay 
superior to 1s, I also fax in the same conditions, so to my experience 
faxes do work with delays far superior to 75ms.

Am I missing something ?
Pedro Caria
On 27/fev/2005, at 17:10, Lee Howard wrote:
On 2005.02.27 08:34 Martijn van Oosterhout wrote:
Hi,
I read it and found it very enlightening. I do have one question
regarding Modems don't like relativity. It says modems need a
constant delay; is there a limit to what it can handle. For example,
would it be possible to configure a jitterbuffer right at the endpoint
before the fax to put a constant delay of 1 second relative to the
sender. This should be enough time to weed out any jitter. Basically,
fix the jitterbuffer so the delay is constant. If a fax can handle a
constant delay of up to a second you're home.
Fax cannot handle a one-second delay.  As Steve mentions in the 
article, per-spec fax has some timings (particularly silence in 
direction switching) set at 75 ms +/- 20 ms.  So if the delay gets 
much larger than 75 ms, then there's likely to be trouble.  Now, some 
fax machines may tolerate larger delays, but that tolerance is beyond 
the spec, and thus should not be used as a gauge.

Lee.
___
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] HELP NEEDED ASTERISK AND MEDIATRIX 1102

2005-02-25 Thread Pedro
Do yourself a favor and get a Sipura SPA-2100 - much easier to
configure and the quality is better than the Mediatrix unit.  First of
all - do you have the Mediatrix Unit Manager software?  If not,
configuration will be nearly impossible.  Secondly, you will need to
configure the sip ports on the mediatrix to include asterisk as the
realm.  The other fields are pretty self explanatory (username,
password, etc.).  You will also want to turn off silence suppression
as it is on by default.

- Pedro


On 25 Feb 2005 20:07:04 +0100, Edward Banfa [EMAIL PROTECTED] wrote:
 Hello all,
 
 Hi I would like to know how to configure a Mediatrix 1102 box to work
 with my asterisk box. I have analog phones that i would like to connect
 to my Mediatrix box and then connect the Mediatrix box to my asterisk
 box. My main problems come from the fact that I have limited experience
 with usiing the two (asterisk and the mediatrix). I know how to use
 sip.conf , but I am lost when it comes to mediatrix specific
 configuration. I have search the archives but i have not gotten any
 thing specific.
 I would really appreciate any help that can be rendered to set me in the
 right path. I am desperate here.
 Thank you all in advance
 
 Edward
 
 ___
 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] IAX channel unable to create

2005-02-21 Thread Pedro
First off - 

change:
exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])

to:
exten = _2000,1,Dial(IAX2/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])

On Mon, 21 Feb 2005 13:00:39 -0500, kurt x [EMAIL PROTECTED] wrote:
 I have two * boxes running two differnet versions of *.
  Box A is running:
 
 Asterisk CVS-HEAD-07/14/04-16:28:29 built by
 [EMAIL PROTECTED] on a i686 running Linux
 
 Box B is running:
 
 Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD
 
 I can make a IAX call from B to A but not from A to B.
 When I try to make a call from A to B I get these messages:
 
 Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_request: No
 channel type registered for 'IAX'
 Feb 21 12:48:12 NOTICE[-1233155152]: app_dial.c:696 dial_exec: Unable
 to create channel of type 'IAX'
 Feb 21 12:48:14 WARNING[-1116300368]: chan_sip.c:673 retrans_pkt:
 Maximum retries exceeded on call
 [EMAIL PROTECTED]
 for seqno 1 (Non-critical Response)
 
 My box A iax.conf:
 [general]
 port=5036
 bindport=5036
 bandwidth=low
 allow=ulaw
 disallow=lpc10
 jitterbuffer=no
 tos=lowdelay
 
 [slave]
 type=friend
 secret=4435
 context=voice-mail
 defaultip=192.168.2.232
 qualify=yes
 
 My Box A extension.conf
 [voice-mail]
 exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])
 
 My box B iax.conf
 [general]
 port=5036
 bindport=5036
 bandwidth=low
 allow=ulaw
 disallow=lpc10
 tos=lowdelay
 
 [master]
 type=friend
 secret=4435
 context=home
 defaultip=192.168.1.2
 qualify=yes
 
 My Box B extension.conf
 [home]
 exten = _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED])
 
 Thanks in advance
 
 Kurt
 ___
 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] I have a odd question...

2005-02-19 Thread Pedro
If you use the MySQL CDR add-on, you could just query the CDR DB for
the numbers you are tracking.  No need to add anything fancy.


On Sat, 19 Feb 2005 21:42:31 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 
 
 Hi all.
 
 I am going to do a simple voting application for a radiostation.
 
 The idea is to have listeners call in to vote on songs.
 
 What I want to do is to take a phonenumer for each song and present the
 result on a simple webpage.
 
 Eg.
 
 To vote on song number one, call 555-
 
 To vote on song number two, call 555-  etc etc.
 
 When the listener calls in, a playback tells him: Thank you for voting
 on song number one.
 
 And the numbers of calls on each number are presented on a webpage, or
 in a textfile, easy for the showhost to see.
 
 How do I do this the simplest way ?
 
 I have a lot on phonenumbers that I can use, so that is not the problem.
 
 Shoud I execute some kind of script for each caller that increases the
 numbers in a textfile ?  Or how should I do ?
 
 My programmingskills aren't the best, so I would be greatful for any
 help I can get.
 
 /Regards Mike.
 
 PS. Please answer offlist if possible..
 
 ___
 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] Vonage, broadvoice et al

2005-02-18 Thread Pedro
Vonage, to my knowledge, does not let you connect your own SIP device
to their service.  They provide their own IAD.

As for Broadvoice, I know people that have successfully deployed
asterisk with many people sharing the same account.

- Pedro


On Fri, 18 Feb 2005 17:54:38 +0800, el Flynn [EMAIL PROTECTED] wrote:
 Hi all,
 
 I'm just wondering about these VoIP services -- do you have to sign up one
 account -per- client that will be using the service? I've got multiple
 extensions behind my Asterisk box, and I want to be able to allow all my staff
 to place calls via the provider.
 
 So if I sign up for one account, will multiple users behind my Asterisk box be
 able to make calls, using that same account, at the same time? Or do these
 providers typically only allow one call to be in place at any point in time?
 
 Thanks in advance.
 
 Flynn
 
 ___
 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] Sipura g729 call quality to PSTN

2005-02-18 Thread Pedro
Rich - thanks!  Glad I am not the only one seeing this :)

Would be very interested in your results.  No problems that I see yet
with these settings.


On Fri, 18 Feb 2005 10:21:08 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
   That does not sound right at all. The difference between the two Time=
   values should have been 10 (milliseconds).
  
   Did you reboot the Sipura after making the change? There are some values
   in the Sipura that don't take effect until after the next reboot; I don't
   have a clue whether this happens to be one of them.
 
  Yes - sipura was rebooted.  Actually, the changes did seem to take
  affect even before the reboot (verified by call quality improvement
  and ethereal traces).
 
  So in your opinion, instead of 80, it should be a difference of 10?
  If so - then you are saying that the timestamp is in miliseconds?
 
  I am as puzzled as you - really does not seem logical, but call
  quality is finally decent and it does not seem to bother asterisk at
  all.  Do you see any potential problems with this?
 
 I did a fair amount of experimenting this morning using a spa3000 with
 g711 and g729 codecs. I'm more confused now then ever. I also used
 ethereal to inspect timestamps, etc.
 
  spa3k(fxs) - asterisk - IAX(ITSP) - pstn net - analog phone
 
 The spa3k is running v2.0.13(GWg) firmware, and * is CVS-HEAD-02/13/05.
 
 The spa3k had a default RTP Packet Size of .030 (30 milliseconds) even
 though the User Manual indicated that 20 milliseconds is the default.
 Asterisk config is default at 20 milliseconds.
 
 I changed the spa3k rtp from .030 seconds, to .020 seconds for
 consistency. Audio quality seemed to be better when using g711.
 
 Regardless of whether I used g711u or g729, the rtp timestamps were
 always 160 difference between consequtive packets (as observed by
 ethereal).
 
 Changing the spa3k rtp to .010 seconds yielded timestamps that were
 always 80 difference between consequtive packets (same as you
 observed). However, * - spa3k continued to have 160 difference.
 Audio quality seemed to improve another step, and the occasional
 echo that we heard seemed to disappear. Pure guess is the smaller
 rtp size is impacting the jitter buffer and/or echo canceller in
 the spa3k. I'm going to run with these settings for a while to see
 what the longer term impact/stability might be.
 
 Rich
 

___
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] CODEC g723, g729, g711

2005-02-18 Thread Pedro
Make sure you have the proper licenses to use the codecs:

g729
http://www.digium.com/index.php?menu=asterisk_g729

g723
http://www.dspg.com/technology/LicensePricing.html


On Fri, 18 Feb 2005 18:17:58 -0800, Nitesh Divecha
[EMAIL PROTECTED] wrote:
 Hello All,
 
 Any one has success with codec g723  g729?
 I am having extremely hard time to setup this codec.
 The only codec worked is g711a/u.
 
 If I set g723  g729 as first and second choice codec in my sip.conf, VM and
 MeetMe stop working.
 
 Sip.conf
 
 [general]
 port = 5060   ; Port to bind to (SIP is 5060)
 bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
 disallow=all
 ;allow=g273
 ;allow=g729
 allow=ulaw
 allow=alaw
 
 #include sip_nat.conf
 #include sip_additional.conf
 
 I am using Snom 220/200 and all are set to use g729.
 
 Thank you,
 
 Nitesh
 
 ___
 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] Sipura g729 call quality to PSTN

2005-02-17 Thread Pedro
Actually - jitter does not seem to be the issue (sound is not garbled
and does not drop out, it was just very low and fuzzy/staticy when
not set to 10 ms).

It is weird that I have to drop to 10ms, but I have tested some more
and the general consenses from the people I have called said it sounds
fine now with 10ms setting.

Thanks for your help though.

Here is the result set from the ethereal trace using 10ms (RTP stream
sent from Sipura to asterisk):

RTP  Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604121
RTP  Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604201

As you can see there is now a difference of 80 between the Time stamps
 (now to sound dumb, but it would be 80 what?)


On Wed, 16 Feb 2005 19:42:19 -0700, Keith Burns
[EMAIL PROTECTED] wrote:
 Hmmm, that worked?
 
 Interesting that you can change the sample size to 10ms since the standard
 is 20ms that most people don't go below. I know you *can* do below 20 but if
 you are doubt the technical ability of the box it seems strange they are
 capable of that.
 
 This seems to smack of bad de-jitter buffers on the egress gateway... are
 you receiving 20ms sampled RTP ?
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Pedro
  Sent: Wednesday, February 16, 2005 3:20 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
 
  FYI - Seems the latest firmware in conjunction with changing the
  packet size to 10ms improved the call quality to usable.  The Cisco
  7960 is stell superior, but now at least the SPA-2100 is acceptable
  (and with 2 working g729 channels including 3-way calling).
 
 
  On Wed, 16 Feb 2005 15:44:58 -0500, Pedro [EMAIL PROTECTED]
 wrote:
   Forgot to mention that when I set the RTP Packet Size to 20ms that the
   difference was 160 (like the Cisco) but call quality was much worse.
  
  
   On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED]
 wrote:
Thanks for the suggestion.  Changing the RTP Packet Size in the Sipura
to 40ms did improve the call quality slightly, but still well below
par compared to the Cisco 7960.
   
In my ethereal captures, I did notice something interesting.  While
the RTP stream from the Cisco to asterisk seemed to have a 160
diffference in timestamps, the Sipura showed a 320 difference:
   
Cisco:
RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094,
  Time=40666896
RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095,
  Time=40667056
   
Sipura:
RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784,
  Time=434932771
RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784,
  Time=434933091
   
   
On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns
[EMAIL PROTECTED] wrote:
 What is your sample size?

 I believe the 7960 supports 40ms (2 samples) per packet by default.

 Do you have an ethereal trace? Look at the timestamps between RTP
 packets if
 you can't see/modify this setting.


  -Original Message-
  From: [EMAIL PROTECTED]
 [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Pedro
  Sent: Tuesday, February 15, 2005 6:30 PM
  To: Jeffrey Chan
  Cc: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
 
  Actually the SPA-2100 supports 2 g729 channels which is why I
 bought
  it.  Unfortunately, the call quality is just as poor on the 2100
 as it
  is on the 2000.
 
  - Pedro
 
 
  On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan
  [EMAIL PROTECTED]
  wrote:
Is it just a bad implementation of g729 compression with the
 Sipura
  product line?
 

That would be my guess too . why SPA-2000 supports G729 for one
   channel only? no enough CPU power to code/decode G.729 for two
   channels?
  
   Jeffey
  
   www.mutualphone.com
  
  
   On Tue, 15 Feb 2005 16:31:59 -0500, Pedro
 [EMAIL PROTECTED]
 wrote:
uggg.
   
Is anyone out there having any luck with the SPA-2000 or
 SPA-2100
using the g729 codec with decent call quality?
   
   
On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler
  [EMAIL PROTECTED]
 wrote:

 On Feb 14, 2005, at 1:25 PM, Pedro wrote:

 
  Is it just a bad implementation of g729 compression with
 the
 Sipura
  product line?
 

 That would be my guess.

 -mark

 --
 Mark Eissler, [EMAIL PROTECTED]
 Mixtur Interactive, Inc. [EMAIL PROTECTED] 
 http://www.mixtur.com


___
Asterisk-Users mailing list
Asterisk

Re: [Asterisk-Users] Sipura g729 call quality to PSTN

2005-02-17 Thread Pedro
 That does not sound right at all. The difference between the two Time=
 values should have been 10 (milliseconds).
 
 Did you reboot the Sipura after making the change? There are some values
 in the Sipura that don't take effect until after the next reboot; I don't
 have a clue whether this happens to be one of them.

Yes - sipura was rebooted.  Actually, the changes did seem to take
affect even before the reboot (verified by call quality improvement
and ethereal traces).

So in your opinion, instead of 80, it should be a difference of 10? 
If so - then you are saying that the timestamp is in miliseconds?

I am as puzzled as you - really does not seem logical, but call
quality is finally decent and it does not seem to bother asterisk at
all.  Do you see any potential problems with this?
___
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] Sipura g729 call quality to PSTN

2005-02-16 Thread Pedro
Thanks for the suggestion.  Changing the RTP Packet Size in the Sipura
to 40ms did improve the call quality slightly, but still well below
par compared to the Cisco 7960.

In my ethereal captures, I did notice something interesting.  While
the RTP stream from the Cisco to asterisk seemed to have a 160
diffference in timestamps, the Sipura showed a 320 difference:

Cisco: 
RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896
RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056

Sipura:
RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771
RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091


On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns
[EMAIL PROTECTED] wrote:
 What is your sample size?
 
 I believe the 7960 supports 40ms (2 samples) per packet by default.
 
 Do you have an ethereal trace? Look at the timestamps between RTP packets if
 you can't see/modify this setting.
 
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Pedro
  Sent: Tuesday, February 15, 2005 6:30 PM
  To: Jeffrey Chan
  Cc: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
 
  Actually the SPA-2100 supports 2 g729 channels which is why I bought
  it.  Unfortunately, the call quality is just as poor on the 2100 as it
  is on the 2000.
 
  - Pedro
 
 
  On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED]
  wrote:
Is it just a bad implementation of g729 compression with the Sipura
  product line?
 

That would be my guess too . why SPA-2000 supports G729 for one
   channel only? no enough CPU power to code/decode G.729 for two
   channels?
  
   Jeffey
  
   www.mutualphone.com
  
  
   On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED]
 wrote:
uggg.
   
Is anyone out there having any luck with the SPA-2000 or SPA-2100
using the g729 codec with decent call quality?
   
   
On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED]
 wrote:

 On Feb 14, 2005, at 1:25 PM, Pedro wrote:

 
  Is it just a bad implementation of g729 compression with the
 Sipura
  product line?
 

 That would be my guess.

 -mark

 --
 Mark Eissler, [EMAIL PROTECTED]
 Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com


___
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 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] Sipura g729 call quality to PSTN

2005-02-16 Thread Pedro
Forgot to mention that when I set the RTP Packet Size to 20ms that the
difference was 160 (like the Cisco) but call quality was much worse.


On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote:
 Thanks for the suggestion.  Changing the RTP Packet Size in the Sipura
 to 40ms did improve the call quality slightly, but still well below
 par compared to the Cisco 7960.
 
 In my ethereal captures, I did notice something interesting.  While
 the RTP stream from the Cisco to asterisk seemed to have a 160
 diffference in timestamps, the Sipura showed a 320 difference:
 
 Cisco:
 RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896
 RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056
 
 Sipura:
 RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771
 RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091
 
 
 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns
 [EMAIL PROTECTED] wrote:
  What is your sample size?
 
  I believe the 7960 supports 40ms (2 samples) per packet by default.
 
  Do you have an ethereal trace? Look at the timestamps between RTP packets if
  you can't see/modify this setting.
 
 
   -Original Message-
   From: [EMAIL PROTECTED] [mailto:asterisk-users-
   [EMAIL PROTECTED] On Behalf Of Pedro
   Sent: Tuesday, February 15, 2005 6:30 PM
   To: Jeffrey Chan
   Cc: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
  
   Actually the SPA-2100 supports 2 g729 channels which is why I bought
   it.  Unfortunately, the call quality is just as poor on the 2100 as it
   is on the 2000.
  
   - Pedro
  
  
   On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED]
   wrote:
 Is it just a bad implementation of g729 compression with the Sipura
   product line?
  
 
 That would be my guess too . why SPA-2000 supports G729 for one
channel only? no enough CPU power to code/decode G.729 for two
channels?
   
Jeffey
   
www.mutualphone.com
   
   
On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED]
  wrote:
 uggg.

 Is anyone out there having any luck with the SPA-2000 or SPA-2100
 using the g729 codec with decent call quality?


 On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED]
  wrote:
 
  On Feb 14, 2005, at 1:25 PM, Pedro wrote:
 
  
   Is it just a bad implementation of g729 compression with the
  Sipura
   product line?
  
 
  That would be my guess.
 
  -mark
 
  --
  Mark Eissler, [EMAIL PROTECTED]
  Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
 
 
 ___
 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 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] Sipura g729 call quality to PSTN

2005-02-16 Thread Pedro
FYI - Seems the latest firmware in conjunction with changing the
packet size to 10ms improved the call quality to usable.  The Cisco
7960 is stell superior, but now at least the SPA-2100 is acceptable
(and with 2 working g729 channels including 3-way calling).


On Wed, 16 Feb 2005 15:44:58 -0500, Pedro [EMAIL PROTECTED] wrote:
 Forgot to mention that when I set the RTP Packet Size to 20ms that the
 difference was 160 (like the Cisco) but call quality was much worse.
 
 
 On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote:
  Thanks for the suggestion.  Changing the RTP Packet Size in the Sipura
  to 40ms did improve the call quality slightly, but still well below
  par compared to the Cisco 7960.
 
  In my ethereal captures, I did notice something interesting.  While
  the RTP stream from the Cisco to asterisk seemed to have a 160
  diffference in timestamps, the Sipura showed a 320 difference:
 
  Cisco:
  RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896
  RTP  Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056
 
  Sipura:
  RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771
  RTP  Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091
 
 
  On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns
  [EMAIL PROTECTED] wrote:
   What is your sample size?
  
   I believe the 7960 supports 40ms (2 samples) per packet by default.
  
   Do you have an ethereal trace? Look at the timestamps between RTP packets 
   if
   you can't see/modify this setting.
  
  
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Tuesday, February 15, 2005 6:30 PM
To: Jeffrey Chan
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN
   
Actually the SPA-2100 supports 2 g729 channels which is why I bought
it.  Unfortunately, the call quality is just as poor on the 2100 as it
is on the 2000.
   
- Pedro
   
   
On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED]
wrote:
  Is it just a bad implementation of g729 compression with the Sipura
product line?
   
  
  That would be my guess too . why SPA-2000 supports G729 for one
 channel only? no enough CPU power to code/decode G.729 for two
 channels?

 Jeffey

 www.mutualphone.com


 On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED]
   wrote:
  uggg.
 
  Is anyone out there having any luck with the SPA-2000 or SPA-2100
  using the g729 codec with decent call quality?
 
 
  On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED]
   wrote:
  
   On Feb 14, 2005, at 1:25 PM, Pedro wrote:
  
   
Is it just a bad implementation of g729 compression with the
   Sipura
product line?
   
  
   That would be my guess.
  
   -mark
  
   --
   Mark Eissler, [EMAIL PROTECTED]
   Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
  
  
  ___
  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 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] Sip phones how to dial a # sign?

2005-02-15 Thread Pedro
I have had this same problem.  The only way I know is to disable
transfers in asterisk.  You can still use the transfer control in your
SIP device.  Of course this does not work with call parking.  I would
be very interested in a solution that does not require disabling of
transfers in asterisk as well.

Pedro


On Tue, 15 Feb 2005 09:52:56 +0100 (CET), Remco Barende
[EMAIL PROTECTED] wrote:
 Hi list!
 
 I have some sip phones and Sipura ATA 2000's. However after dialling a
 number I need to dial a # to control a device.
 
 When I dial # Asterisk kicks in and puts the call on hold. How can I
 change this?
 
 Thx!!
 Remco
 ___
 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] Sixtel.net / IAX.CC - Vanity Toll-Free Number

2005-02-15 Thread Pedro
Same boat here.

Actually got someone on AOL instant messenger yesterday.  Their
response as follows when asked how long it will take to get our 800
number:

[15:11] sixtel9: it's in the works 

any time frame?
[15:14] sixtel9: not specifically, we switched carriers so we're
dealing w/ some issues

just need to know if it will be weeks/months/ or days
[15:21] sixtel9: days

- Pedro

On Tue, 15 Feb 2005 09:41:12 -0500 (EST), Paul Dugas
[EMAIL PROTECTED] wrote:
 On Tue, February 15, 2005 9:27 am, Rob Risner said:
  I'm just wondering, how long should a vanity number transfer really take?
 
 No help here, just posting a me too to warn others.  Friday was 10 days
 for me.  No happy to hear you've waited much longer with the same result.
 Can never raise them on the phone.  They take days to respond to the
 ticket and are rather terse when they actually do.
 
 Not pleased at all.
 
 Paul
 
 --
 Paul A. DugasDugas Enterprises, LLC
 [EMAIL PROTECTED]1711 Indian Ridge Drive
 p:404-932-1355  f:770-516-4841   Woodstock, GA 30189-6856 USA
 ___
 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] Sip phones how to dial a # sign?

2005-02-15 Thread Pedro
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but still keep the T in
the dial plan?  We have clients that need to check external voicemail
systems that require the use of the # sign, but still want to have the
call parking feature.


On Tue, 15 Feb 2005 06:54:23 -0700, Michael Welter [EMAIL PROTECTED] wrote:
 Remco Barende wrote:
  Hi list!
 
  I have some sip phones and Sipura ATA 2000's. However after dialling a
  number I need to dial a # to control a device.
 
  When I dial # Asterisk kicks in and puts the call on hold. How can I
  change this?
 
 Do you have the T in your Dial statment? Remove the T and try it.
 ___
 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] Clarification on Fax capability?

2005-02-15 Thread Pedro Miguel de Sousa Caria
I've been trying this for a while and I have been unable to get a 
reliable connection betwen two Zaptel FXS interfaces, so the bridging 
does afect data transfer.

Anybody got some tunning tips to get this to work ?
I'm using a dual PIII with a ServerWorks Chipset, two TDM cards (8xFXS) 
and a Fritz Capi to connect to my telecom provider.

I can send faxes with some success, but receiving rate of success is 
less than 30%.

Fax information for Asterisk is difficult to come by is everybody using 
spandsp's way ?

Thx
Pedro Caria
On 15/fev/2005, at 15:05, Rich Adamson wrote:
On Tue, February 15, 2005 7:48 am, Rich Adamson said:
 2) simply switching a fax call through * to a tip/ring interface of
some sort that has an attached traditional fax machine.
Does the codec issue with #2 still apply if the incoming fax call is 
on a
Zaptel FXO interface?  Is the codec used when connecting two channels 
on
the same zaptel card or does the native bridg[ing] bypass that?
Funny that you should ask. I just finished testing it using the
faxdetect=incoming method, redirecting the call to a Cisco ata186.
The incoming call arrived on a TDM fxo port. Received the fax header,
but the remainder of the page was blank. The sender received a message
indicating a failure.
Best guess... the tdm driver problem is impacting the ability to send
the fax tones reliably even with g711. Its very likely to be the
interrupt latency and/or pci bus problem on this particular system.
___
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


  1   2   >