Re: [Asterisk-Users] Call forward...

2005-05-18 Thread adria vidal 
El 18/05/2005, a las 11:42, Mark Benson escribió:
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1", 
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable 
to create channel of type '' (cause 66)
 == Everyone is busy/congested at this time (1:0/0/1)

The call then drops into voicemail...

Maybe you have to erase the " in your Trunk variable ?

··
Adrià Vidal
...

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Re: [Asterisk-Users] Implant GIPS's codec to Asterisk

2005-03-30 Thread adria vidal 
El 30/03/2005, a las 7:40, Dominic Lu escribió:
Hello,
If purchase the codec from GIPS, how difficult it is to implant it in 
Asterisk? What the cost will be?

Our company has two Asterisk, one in headquarter and the other in 
branch office. We only need the communication between them. We are not 
satisfied with current codec either in bandwidth usage or voice 
quality. Since Skype really impress us in voice quality, so this kind 
of idea is generated.  

BR, Dominic

You are talkina about GIPS ILBC ?  
http://www.globalipsound.com/products/iLBCfreeware.phpThere is a free 
ILBC codec  http://www.ilbcfreeware.org/
 ILBC is compiled by default by asterisk. My friends usually say sound 
quality in asterisk is better than Skype one, i've heard that Skype use 
a modified version of ILBC


··
Adrià Vidal
 ...

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Re: [Asterisk-Users] Re: Asterisk and XLite on same machine (OSX)?

2005-03-29 Thread adria vidal 
why not using a IAX phone, is running great on OS X
http://iaxclient.sourceforge.net/iaxcomm/

··
Adrià Vidal 
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[Asterisk-Users] CAPI- 2 Cards

2005-03-11 Thread adria vidal 
Some suggestion about how detect busy channels in a installation with 2 cards (AVM Fritz)? 
Can't find info about groups in capi channels.  Need to dial out trought some of the 4 avalaible channels.
Better try it with zaphfc ?

Adrià Vidal 
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Re: [Asterisk-Users] "click to dial extension number" functionality ?

2005-02-25 Thread adria vidal 
El 25/02/2005, a las 12:10, Terje Myhre escribió:
By any web-user (ms explorer) to be able to call from a web-page to a  
certain number/extension connected to one specific asterisk.

  
Almost as a web-based “auto-attendant” functionality. 
  
Hence:
1.   surf to the specific web-site
2.  enter the extension digits in a web-interface
3.  get connected – with in- and out-sound through the web-browser
  
Do anyone know what would be the simplest / best way to implement this  
functionality ?

We have developed something similar
http://www.asteriskspain.org/index.php? 
option=com_remository&Itemid=41&func=selectfolder&filecatid=1

you choose an extension put your phone number and asterisk make and  
bridge calls, it's easy to personalize if you want.

··
Adrià Vidal
xpreme.net
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Re: [Asterisk-Users] Snom phone hint exten question

2005-02-21 Thread adria vidal 
El 21/02/2005, a las 12:30, James Bean escribió:
Still doesn't work, I dialed in an outside line and picked up the
receive on extension 691, yet the light on the snom phone did not come
on. I dialed out of extension 691 to an outside line, yet still the
light did not come on.
Snom190 has firmware 3.56m the button is set to Destination 691

Be sure to reboot the snom after every change, fooled with it a little 
bit too.
But get it woorking now.

Atentament.
··
Adrià Vidal 
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Re: [Asterisk-Users] Asterisk 1.0.5

2005-01-24 Thread adria vidal 
El 24/01/2005, a las 4:34, Russell Bryant escribió:
Hello everyone,
As you know, we released Asterisk 1.0.4 earlier this week.  However,
there was a callerid bug in chan_zap that has caused us to go ahead and
make another release.  Asterisk 1.0.5 is available at all of the usual
locations.
I'm sorry for any inconvenience this may cause.

What about zaptel and libpri? are they ok, can continue running the 
versions i have now?

Adrià Vidal 
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[Asterisk-Users] h323 compilation problem

2005-01-19 Thread adria vidal 
Someone have had good luck compilig h323 into YDL?
first thinked was a bug in code but "twisted" said it is
" wierd - isn't that the recursive pthread lib? If so, do you have the 
kernel development headers/libs installed?"
I've instaled kernel source, what more can i do? any help would be very 
wellcome, i've googled everywhere but can't find nothing about it.


make install
new error
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT 
-D_GNU_SOURCE -O6 -fsigned-char -DZAPTEL_OPTIMIZATIONS 
-DASTERISK_VERSION=\"CVS-NHEAD-01/17/05-19:15:56\" 
-DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\" 
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" 
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" 
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" 
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" 
-DASTMODDIR=\"/usr/lib/asterisk/modules\" 
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN 
-Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI 
-DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_h323.o chan_h323.c
gcc -shared -Xlinker -x -o chan_h323.so chan_h323.o h323/libchanh323.a 
-ldl -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib 
-lh323_linux_x86_r -L/usr/lib -lcrypto -lssl -lexpat
/usr/bin/ld: cannot find -lpt_linux_x86_r
collect2: ld returned 1 exit status
make[1]: *** [chan_h323.so] Error 1
make[1]: Leaving directory `/mnt/60gbide2/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk]#

Adrià Vidal
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[Asterisk-Users] oh323 compile problem still

2005-01-13 Thread adria vidal 
Followed instructions from these old post, CVS updated my asterisk too,  
edites makefile... but

--
Get oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/ 
openh323-Janus_patch4-src-tar.gz
Get pwlib from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/ 
pwlib-Janus_patch4-src-tar.gz
Get asterisk-oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/ 
asterisk-oh323-0.7.tar.gz

Untar the files
#tar zxvf openh323-Janus_patch4-src-tar.gz
#tar zxvf pwlib-Janus_patch4-src-tar.gz
#tar zxvf asterisk-oh323-0.7.tar.gz
Install Pwlib
#cd pwlib
#./configure && make clean && make opt && make install && ldconfig
Patch and Install OpenH323
#cd openh323
#patch -p1 < ../asterisk-oh323-0.7/openh323_1.13.5-make.patch
#./configure && make clean && make opt && make install && ldconfig
---
get this compile error. Please help, is my third time trying to install  
h323 support into *
Some suggestion?

chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2581: structure has no member named `cid'
chan_oh323.c:2587: structure has no member named `cid'
chan_oh323.c:2588: structure has no member named `cid'
chan_oh323.c:2589: structure has no member named `cid'
chan_oh323.c:2594: structure has no member named `cid'
chan_oh323.c:2595: structure has no member named `cid'
chan_oh323.c:2596: structure has no member named `cid'
chan_oh323.c:2598: structure has no member named `cid'
chan_oh323.c:2598: structure has no member named `cid'
chan_oh323.c:2643: structure has no member named `cid'
chan_oh323.c:2644: structure has no member named `cid'
chan_oh323.c:2645: structure has no member named `cid'
chan_oh323.c:2646: structure has no member named `cid'
chan_oh323.c:2647: structure has no member named `cid'
chan_oh323.c:2648: structure has no member named `cid'
chan_oh323.c: In function `load_module':
chan_oh323.c:5201: warning: passing arg 4 of `ast_channel_register'  
from incompatible pointer type
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory  
`/mnt/60gbide2/usr/src/asterisk-oh323-0.7.0/asterisk-driver'
make: *** [subdirs_build] Error 1
[EMAIL PROTECTED] asterisk-oh323-0.7.0]#

Adrià Vidal
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Re: [Asterisk-Users] chan_capi-0.3.5 error 127

2005-01-12 Thread adria vidal 
El 12/01/2005, a las 15:36, Vincent Guidoux escribió:
 
Hi,
I have a problem for install chan_capi
 
My pc: Suse 9.1, with asterisk current stable en cvs
 
I have download 
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
 
And the path from 
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
 
And patch the chan_capi
 
# cd chan_capi-0.3.5
# patch –p1 < chan_capi-0.3.5-patch.diff
# make    /for install
 
My error :
 
Make: gcc-2.95 :Command not found
Make: *** [chan_capi.o] Error 127
 
If
 
# make CC=gcc-3.3.3    /because I have gcc-3.3.3
 
Error:
 
Make: gcc-2.95 :Command not found
Make: *** [chan_capi.o] Error 127
 
If
 
# make CC=gcc
In file included from /usr/include/linux/kernelcapi.h:13,
    from /usr/include/linux/capi.h:18,
    from chan_capi.c:35:
/usr/include/linux/list.h:604:2: warning: #warning “don’t include 
kernel headers in userspace”
chan_capi.c: In function ‘capi_new’:
chan_capi.c:1076: error: structure has no member named ‘cid’
chan_capi.c:1077: error: structure has no member named ‘cid’
chan_capi.c: In function ‘capi_handle_dtmf_fax’:
chan_capi.c:1189: error: structure has no member named ‘cid’
chan_capi.c: In function ‘pipe_msg’:
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c:1764: error: structure has no member named ‘cid’
chan_capi.c: In function ‘load_module’:
chan_capi.c:2843: warning: passing arg 4 of ‘ast_channel_register’ 
from incompatible poiner tpe
make: *** [chan_capi.o] Error 1
 
 
I don’t know if the problem from gcc or oder, if you can help me, 
thanks soo much!


coment lines into makefile looking for an specific gcc, and it will 
compile fine.

Adrià Vidal
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Re: [Asterisk-Users] hint extension and Snom phones - CVS or stable?

2004-12-22 Thread adria vidal 
El 22/12/2004, a las 1:51, Eric Wieling aka ManxPower escribió:
No.  Hint is not supported in 1.0.x.  Only in CVS-HEAD developement 
version of Asterisk.

--Eric
running fine for my in 1.0.3 release and snom 190
adrià
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Re: [Asterisk-Users] Fw: pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 1)

2004-12-20 Thread adria vidal 
El 20/12/2004, a las 9:25, [EMAIL PROTECTED] escribió:
 I have added a sip user in sip.conf. user name is 819,context is c819.
and I have added the follows rows in extension.conf. like
[c819]
  exten => 1,1,Answer
  exten => 1,2,SetVal(voicemail=${exten})
  exten => 1,3,Dial(SIP/${voicemail})
It appear a error message(pbx.c:1279 pbx_extension_helper: No 
application 'SetVal' for extension (c819, 1, 2)) when i dial 1 from 
819.

The version of asterisk is 1.0.3
Please help me. Thank a lot.  

Try with SetVar

··
Adrià Vidal |
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[Asterisk-Users] Zap gives no ring to the caller...

2004-11-29 Thread adria vidal 
I have a E1 conected to asterisk all zap channels are ok, but when calls come into Asterisk caller don't hear none ring, the call goes straight into the menu, how can i simulate 2 or 3 rings?

here it is my conf.

exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,NoOp(${CALLERID})

exten => s,4,ResponseTimeout,45
exten => s,5,DigitTimeout,3
exten => s,6,Background(wellcome)


Adrià Vidal 
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[Asterisk-Users] Graststream ATA 286 Caller ID Europe

2004-11-24 Thread adria vidal 

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ?


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[Asterisk-Users] h323 asterisk

2004-11-04 Thread adria vidal 
Using las CVS Head can't compile channels/h323
have downloaded some diferent pwlib and openh323 because can't find the 
exact ones called into README, have some suggestion?

/usr/src/pwlib/include/ptlib/pfactory.h:181: ISO C++ forbids 
declaration of `
   PMutex' with no type
/usr/src/pwlib/include/ptlib/pfactory.h:181: parse error before `&' 
token
/usr/src/pwlib/include/ptlib/pfactory.h:183: 'PMutex' is used as a 
type, but is
   not defined as a type.
/usr/src/pwlib/include/ptlib/pfactory.h:193: parse error before `>' 
token
/usr/src/pwlib/include/ptlib/pfactory.h:198: template declaration of 
`typedef
   _Abstract_T Abstract_T'
/usr/src/pwlib/include/ptlib/pfactory.h:198: confused by earlier 
errors, bailing out
{standard input}: Assembler messages:
{standard input}:186: Error: symbol `readerCount' is already defined
{standard input}:192: Error: symbol `writerCount' is already defined
make: *** [ast_h323.o] Error 1

adrià vidal
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Re: [Asterisk-Users] ZTdummy

2004-11-02 Thread adria vidal 
El 01/11/2004, a las 21:54, Paul Rodan escribió:
It's picky about what USB controller you have. It refused to work on my
server because it had the wrong kind of USB controller, go figure.
So I used zaprtc and it worked fine. If you have problems with 
zrdummy, let
me know and I'll see if I can help with zaptelrtc. The trickiest part 
is to
make sure you don't have the Real Time Clock (rtc) compiled in the 
kernel.

Can't find info into the Wiki about how to intall Zaprtc ? Ztdummy is 
not working with my server neither.
 can help me a little bit?

··
Adrià Vidal
...

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Re: [Asterisk-Users] Vonage Softphone--outbound calls work, inbound do not

2004-10-25 Thread adria vidal 
El 25/10/2004, a las 15:53, Richard Branham escribió:
register => 
:@sphone.vopr.vonage.net:5061/
Maybe your incoming calls are going to a non existent number in your 
system  ???

try
register => 
:@sphone.vopr.vonage.net:5061/

··
Adrià Vidal
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Re: [Asterisk-Users] 827-4V voice ports, asterisk and hookflash

2004-10-22 Thread adria vidal 
El 22/10/2004, a las 8:55, Justin Hawkins escribió:
Hi folks,
I have setup my Cisco 827-4V to talk to asterisk, with success. I can
make and receive calls. The world is good.
I am wondering however about extra features - putting people on hold
and parking calls and so on. These features seem to require me to use
hookflash.

I have the same one, no problem with parking or transfering calls with 
asterisk, haven't tried put on hold...
Feel free to contact offline if you want, maybe can share configs.

··
Adrià Vidal
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Re: [Asterisk-Users] Changing the default language

2004-10-13 Thread Adria Vidal 
El 13/10/2004, a las 12:48, ismaelg escribió:
How could I change the default Languaje for Voicemail?
I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have 
a letter and diggits directory too.

Any clue will be appreciated.

Mine is running fine, try it.
exten => 207,1,Dial(SIP/[EMAIL PROTECTED],10,Ttr)
exten => 207,2,SetLanguage,fr
exten => 207,3,Voicemail(${EXTEN})
exten => 207,4,Hangup
Adrià Vidal
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[Asterisk-Users] DID's in Spain

2004-08-02 Thread Adria Vidal 
Someone giving DID for Spain?
Thanks in advance
Adrià Vidal
mailto:adriavidal at telefonica.net
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[Asterisk-Users] Asterisk CVS compile error YDL 3.0.1

2004-07-20 Thread Adria Vidal 
Hi, i'm trying to compile Asterisk under YDL 3.0.1, libpri, zaptel compile ok, but at make install in asterisk give me this error, have an idea because it can be? Thanks in advance.


k\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\"
-DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\
" -DBUSYDETECT_MARTIN  -fPIC  -Wall -Werror -fPIC -O3 -march=ppc -funroll-l
oops -fomit-frame-pointer   -c -o anaFilter.o anaFilter.c
cc1: invalid option `arch=ppc'
make[2]: *** [anaFilter.o] Error 1
make[2]: Leaving directory `/usr/src/asterisk/codecs/ilbc'
make[1]: *** [ilbc/libilbc.a] Error 2
make[1]: Leaving directory `/usr/src/asterisk/codecs'
make: *** [subdirs] Error 1


Adrià Vidal 

[EMAIL PROTECTED]

Re: [Asterisk-Users] PSTN Gateway X101P

2004-07-18 Thread Adria Vidal 
On Jul 18, 2004, at 5:56 PM, Jason Armentrout wrote:
to the extensions.conf
but I am not sure I follow you on the second part, do you want me to 
add

include => outgoing
to my sip.conf file?? I did both of these changes, and I still have 
the same
problem.


must add
include => outgoing
into your extensions.conf file where the sip extensions are defined 
example

[sip]
;
include => fwd
include => iaxtel
include => stanaphone
include => SIPphone
include => fromiaxfwd
include => from-iaxtel
include => stana-incoming
include => parkedcalls
include => outgoing

exten => 100,1,Dial(SIP/100,20,tr)
exten => 100,2,Voicemail,100
exten => 100,3,Hangup


Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN 
[EMAIL PROTECTED]
iChat [EMAIL PROTECTED] | FWD  [EMAIL PROTECTED] | IAXTEL  1700 337 68 
48

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Re: [Asterisk-Users] PSTN Gateway X101P

2004-07-18 Thread Adria Vidal 
try puttin this in extensions.conf
[outgoing]
exten => _0.,1,Dial,Zap/1/${EXTEN:1}
exten => _0.,2,Hangup

and into your siphones extensions definition
[sip]
include => outgoing
Adrià Vidal
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48

On Jul 18, 2004, at 5:12 PM, Jason Armentrout wrote:
1 channels configured.
It appears that I have the driver loaded correctly.
I edited the sample extensions.conf and changed the varible trunk to 
zap/1

Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would be appreciated.
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Re: [Asterisk-Users] Mac OS X installer for Asterisk

2004-07-17 Thread Adria Vidal 
Thanks a lot Benjamin, that's great.
Support for the zaptel drivers would be great! installing the X100P 
card from my linux machine into the OS X one would be incredible

Adrià Vidal
On Jul 17, 2004, at 8:09 PM, Sunrise Ltd wrote:
Anyone who'd like to give this a try, please download the
installer package from here ...
http://www.astmasters.net/stuff/Asterisk.pkg.tgz
to install Asterisk on OSX just double click the package
file.
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Re: [Asterisk-Users] X101P Board UNCONFIGURED

2004-07-16 Thread Adria Vidal 
On 16/07/2004, at 11:16, Holger Schurig wrote:
And the zaptel.conf is in /etc, not in /etc/asterisk?
Thanks a lot was that, editing file in /etc/asterisk not the good one 
for Zaptel.

Dial in and out running now.
Adrià Vidal
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Re: [Asterisk-Users] X101P Board UNCONFIGURED

2004-07-16 Thread Adria Vidal 
On 16/07/2004, at 9:39, Holger Schurig wrote:
a) don't reply to a thread and just change the subject. Instead, start 
a
new thread.
sorry thought was the same.
b) did you run ztcfg?
Yes and i get:

[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
0 channels configured.

Adrià Vidal
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[Asterisk-Users] X101P Board UNCONFIGURED

2004-07-15 Thread Adria Vidal 
Any help setting up a X101P in Spain
zttool show it as UNCONFIGURED  (or in RED when line is out, so the 
card is running ok)


zaptel.conf
loadzone = fr
defaultzone = fr
fxsks=1
zapata.conf
;
; Zapata telephony interface sample configuration file
;
[channels]
;
; X100P plugged into PSTN
;
context=incoming
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
channel => 1
Adrià Vidal
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Re: [Asterisk-Users] *** Asterisk Sunday (hrrm) News: Moving ahead at CVS Warp 5

2004-07-07 Thread Adria Vidal 
On 07/07/2004, at 7:28, Dr. Rich Murphey wrote:
I wish I had access to an OS X system.  I could maintain more of the 
common
*BSD support if so.

Cheers,
Rich

I could open a login account on a OS X system if these can help you.
Adrià Vidal
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