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
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
why not using a IAX phone, is running great on OS X
http://iaxclient.sourceforge.net/iaxcomm/
··
Adrià Vidal
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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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El 25/02/2005, a las 12:10, Terje Myhre escribi:
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
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
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
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
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
El 12/01/2005, a las 15:36, Vincent Guidoux escribi:
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
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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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
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(${CALLERID})
exten =>
Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ?
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El 01/11/2004, a las 21:54, Paul Rodan escribió:
It's picky about what USB controller you have. It refused to work on my
server because it had the wrong kind of USB controller, go figure.
So I used zaprtc and it worked fine. If you have problems with
zrdummy, let
me know and I'll see if I can
El 25/10/2004, a las 15:53, Richard Branham escribió:
register =
VonageNumber:VonagePassword@sphone.vopr.vonage.net:5061/
Maybe your incoming calls are going to a non existent number in your
system ???
try
register =
VonageNumber:VonagePassword@sphone.vopr.vonage.net:5061/
El 22/10/2004, a las 8:55, Justin Hawkins escribió:
Hi folks,
I have setup my Cisco 827-4V to talk to asterisk, with success. I can
make and receive calls. The world is good.
I am wondering however about extra features - putting people on hold
and parking calls and so on. These features seem to
El 13/10/2004, a las 12:48, ismaelg escribió:
How could I change the default Languaje for Voicemail?
I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have
a letter and diggits directory too.
Any clue will be appreciated.
Mine is running fine, try it.
exten =
Someone giving DID for Spain?
Thanks in advance
Adrià Vidal
mailto:adriavidal at telefonica.net
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Hi, i'm trying to compile Asterisk under YDL 3.0.1, libpri, zaptel compile ok, but at make install in asterisk give me this error, have an idea because it can be? Thanks in advance.
k\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\
try puttin this in extensions.conf
[outgoing]
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
and into your siphones extensions definition
[sip]
include = outgoing
Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN
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iChat [EMAIL PROTECTED] | FWD [EMAIL
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
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
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.
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
mailto:[EMAIL PROTECTED]
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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
;
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
mailto:[EMAIL PROTECTED]
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