[EMAIL PROTECTED]:4 Up Playback()
Thanks in advance.
Serge
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[EMAIL PROTECTED]:4 Up Playback()
Thanks in advance.
Serge
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! Serge MESSA OVONO
Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! Mail et son interface révolutionnaire.
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Asterisk to use
the sphinx server.Must i write entirely an eagi
script? And when it's wrote, which changes can i do in
Asterisk or which process does i follow to make sphinx
run with Asterisk?
Best regards!
Hi all
I want to use sphinx2 with asterisk. I install
sphinx but when i type sphinx2-server, i have the
errors below:
ad_oss.c(105): Failed to open audio device(/dev/dsp):
No such device
FATAL_ERROR: "server.c", line 476: ad_open() failed
Thanks for all!
__
Hi all
How can i decrease the speed of festival? It appear
that in festival, the text is read too fast for me
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les t
i'm hear no voice.
What's the problem?
Thanks
Serge
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez
c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (conference, 600,
1)
== Spawn extension (conference, 600, 1) exited
non-zero on 'IAX2/1000-2'
-- Hungup 'IAX2/1000-2'
I install the zaptel module with the ztdummy timer but
the problem still exist.
How c
Hi all
i want to know where i can find french sounds for
asterisk. I don't have any studio to register good
sounds.
Bests regards
Serge
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Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels
: text/plain
On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote:
> Hi all
>
>I'm a newbie in asterisk.I install asterisk
server
> successfully. I configure this server to traverse
NAT.
> Using Xlite clients, i make a call between 2 local
> networks through Internet.Asteri
Hi all
I'm a newbie in asterisk.I install asterisk server
successfully. I configure this server to traverse NAT.
Using Xlite clients, i make a call between 2 local
networks through Internet.Asterisk server is
installed on a host with public IP. client A (in the
LAn A) and client B (in the LAN
No unfortunatly this is not an option, we would never be able to reach them
all with 48 hours.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mercredi 21 décembre 2005 00:04
To: Asterisk Users Mailing List - Non-Commercial Discussi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mardi 20 décembre 2005 23:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IVR and db
Serge Schumacher a écrit :
>Thnx for the f
he stuff I just
wrote below...
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Shawn Porter
Sent: Tuesday, December 20, 2005
4:47 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] I
Hi,
Do you think * could play around 300 voicemenu messages simoultanously?
Regs,
Serge
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,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mardi 20 décembre 2005 20:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IVR and db
Serge Schumacher a écrit :
> Hi,
>
> I ha
Hi,
I have a more general question.
Our group over 5000 employees’ world wide wants to do
a survey for all employees asking them if they are happy with the job, salary,
environment etc…
Can I use an * where people can call a certain phonenumber,
go through voice menues and enter
ed "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-Us
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
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What platform should you suggest to use asterisk ?
I tried with SUSE now all the time but there are too many
problems with the updates.
On is the development platform on which * is developed ?
Regards,
___
Asterisk-Users mailing
hi everybody
I'm a new Asterisker.
I have a very simple configuration : 1 Sip proxy and 2 grandstream 102 in
ethernet with
private adress
sip proxy : 192.168.2.194
ip phone address : 192.168.2.144
192.168.2.195
I want to make a communication between 2 ip phone with the SIP proxy but i
have 2 diff
r
message:
NOTICE[1174440880]: chan_sip.c:7519 handle_request: Registration from
'' failed for 'XXX.XXX.XXX.XXX'
sip.conf:
[Cisco]
...
extensions.conf:
TEST = SIP/[EMAIL PROTECTED]
...
Cisco configuration:
UID0: 150
PWD0: password
UID1: 0
UseLoginID: 0
What is go
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime
First off, stop using HTML email.
What is in your extconfig.conf? What shows when you type "realtime mysql
status"?
-Matthew
- Original Message -
From: "Serge Schumacher" <[EMAIL PROTECTED]&
Console :
*CLI> dial [EMAIL PROTECTED]
No such extension '650' in context
'from-sip'
Extentions.conf
[from-sip]
switch => Realtime/@realtime_ext
extconfig
realtime_ext => mysql,asterisk,extensions_table
res_mysql.conf
[general]
dbhost = 127.0.0.1
dbnam
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime
Serge Schumacher wrote:
> I downloaded latest * stable complile it successfully but when compiling
> the asterisk-addons the res_config_mysql.so is missing.
The stable version of Asterisk does not have Re
I downloaded latest * stable complile it successfully but
when compiling the asterisk-addons the res_config_mysql.so
is missing.
I followed the
instructions on wiki for Realtime.
What did you do wrong ?
Thanx,
___
Asterisk-Use
What control is it ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: vendredi 7 janvier 2005 11:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sip protocol question ...
Hi,
I'm tryinig to debu
I just installed the lastest cvs version of asterisks and
addons but after this the chan_capi doesn’t compile anymore with ne new
header files ?
Anyone an idea ?
Thnx.
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05-01-07 at 03:25 +0100, Serge Schumacher wrote:
> Well I have in asterisk/include/asterisks a lot of .h files if it's what
you
> mean ?
[snip correctly bottom posted reply...]
Iirc during compilation of asterisk-addons it will search in
/usr/include/asterisk for .h files. If you hav
t: Re: [Asterisk-Users] What's wrong with compile
On Fri, 2005-01-07 at 00:12 +0100, Serge Schumacher wrote:
> common.c:1:29: asterisk/logger.h: No such file or directory
> common.c: In function `decode_header':
You don't seem to have asterisk's header files installed in
/usr/inc
Hi,
I checked out asterisk and asterisk-addon into /usr/src, went to
asterisk-addons and did 'make' as usual, but I get
common.c:1:29: asterisk/logger.h: No such file or directory
common.c: In function `decode_header':
I have the zaptel and libpri in the same folder and thy compiled correctly.
Hi,
Jan 6 01:43:09 WARNING[12209]: pbx.c:796
pbx_find_extension: No such switch 'Realtime'
What does this message mean ?
Something wrong with the switch statement in my
extensions.conf or maybe is the module net correctly installed ?
Thnx.
__
rding.
I'll try to fill in some gaps on the Wiki tonight, take a look in the
morning.
http://www.voip-info.org/wiki-Asterisk+manager+API
MATT---
-Original Message-
From: Serge Schumacher [mailto:[EMAIL PROTECTED]
Sent: Monday, January 03, 2005 8:38 PM
To: Asterisk Users Mail
Not yet solved !
Serge Rodrigues
Research & Development
* e-Mail [EMAIL PROTECTED]
* Tel +32 (02) 649 80 89 PERIACTES
* Fax +32 (02) 648 27 22 Av. de l'hippodrome 147
* Webhttp://www.periactes.be B-1050
Hi,
Where can I find a complete * manager api guide, the one one wiki is missing
informations like the monitor function for example,
Thnx
Serge
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How stupid, thanks a lot
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: lundi 3 janvier 2005 01:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme
Serge Schumacher wrote
How stupid, thanks a lot
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: lundi 3 janvier 2005 01:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme
> Can someone see what's wrong here plea
Hi,
Can someone see what's wrong here please ?
I've installed the ztdummy driver to enable meetme, put his in my
extension.conf
exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup
this in my meetme.conf
[rooms]
;
; Usage is conf => confno[,pin]
;
con
Can it be that the MeetMe application is not installed by
default even if there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for
extension (from-sip, 550, 4)
Regards,
___
Asterisk-Users mailing li
: "Serge Schumacher" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Friday, December 31, 2004 8:00 AM
Subject: [Asterisk-Users] IAX users
> Hi,
>
> I do not understand the difference between SIP and IAX, is it onl
Might be related to the musiconhold files using different encoding rates ?
Just an idea, also a newbie :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paid Up
Sent: vendredi 31 décembre 2004 14:01
To: asterisk-users@lists.digium.com
Subject: [Asterisk
Hi,
I do not understand the difference between SIP and IAX, is it only two
different protocols or something more special.
The problem I have is that I've created two users
Aix.conf
register => users1:passwd1
register => user2:passwd2
[user1]
type=user
context=default
secret=passwd1
host=dynam
www.flexcom.lu
- Original Message -
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: 12/5/2004 5:13:07 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing - which program are you using?
> I want to play around with post billing. List of all phone calls, ...
>
> Which program
final
diffusion server. Do you think Asterisk can do this job (I think yes, but not
sure) and what’s the best hardware for 2 simultaneous lines?
Thx a lot.
Serge Rodrigues
Research & Development
*
e-Mail [EMAIL PROTECTED]
( Tel +32 (02) 649 8
final diffusion server. Do you think Asterisk can do
this job (I think yes, but not sure) and what’s the best hardware for 2
simultaneous lines?
Thx a lot.
Serge Rodrigues
Research &
Development
*
e-Mail [EMAIL PROTECTED]
( Tel +32 (02) 649 8
Hi,
What can it be when I can establish a connection between two Softphones but no
voice is transfered ?
thnx
Hugo,
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To UNSUBSCRIBE or update opti
into the DSL modem and let it run
> Asterisk, firewall and routing and NAT and your wireless.
> You will need two network interfaces on the linux box
>
> Outside users just will not be able to get in otherwise.
>
> --- Serge Schumacher <[EMAIL PROTECTED]> wrote:
>
>
t unable to establish a voice call between two SIP
clients.
Someone a clue how it can be solved or... ?
Regs,
Serge
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... It's mistake for kinds.. I'm beginner Linux user, and early half year
back move *asterisk from /usr/bin to /bin ... forgot why.., and now Linux try
start asterisk from old file in /bin ,, Sorry..
Serge.
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Asterisk-Users mailing l
r/lib/asterisk/modules/chan_oh323.so:
undefined symbol: _ZNK13PSoundChannel6IsOpenEvNov 5 01:05:33
WARNING[11040]: loader.c:480 load_modules: Loading module chan_oh323.so
failed!---
Serge.
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Asterisk-Users mailing list
[
Sorry, problem solved, it's my
mistake..
- Original Message -
From:
Serge
To: [EMAIL PROTECTED]
Sent: Thursday, November 04, 2004 9:38
AM
Subject: [Asterisk-Users] res_features.so
Segmentation fault
Have anyone
some idea ?Asterisk - latest
Have anyone
some idea ?Asterisk - latest
cvs,RedHat9==Asterisk
Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf':
Found [chan_modem.so] => (Generic Voice Modem Driver) ==
Parsing '/etc/asterisk/modem.conf': Found == Loading modem
tures.so] => (Call Parking Resource)
== Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
Segmentation fault
Thanks.
Serge.
___
Aster
s/chan_modem.so: undefined
symbol: ast_pthread_create
Nov 3 03:29:06 WARNING[1076231168]: loader.c:374 load_modules: Loading
module chan_modem.so failed!
I have make update from asterisk 07/17/04 cvs...
Please Help...
Regards,
Serge
___
Asterisk-Use
Does anyone working chan_h323 from last cvs with new openh323 1.14.4 and
pwlib 1.7.5 pandora, as a write in 'readme' ?
sound get not, gsm codec after enable in h323.conf don't show in >h.323 show
codec, and after first call asterisk don't make >
Yes, it's work,
Thanks,
But possible don't use Global Var?, due in this situation all other
destinations use this codec, after 1 time global setup. And g729 - limited:(
Regards,
Serge.
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To
return: "No one is available to answer at this time"
Many thanks for your help,
Regards,
Serge.
aky.
I do not want to go through pain of assembling all this stuff together
again. Does anybody know of a Linux distro that would have all these things
running in it "out of the box"? It does not need to have the same
components, but should have the same functi
If you need it I can drop you.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Julian Pawlowski
> Sent: Monday, May 24, 2004 1:26 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Problem receiving a fax with RxFAX
>
> Hello
Hi all,
Unable to find translation path. How to fix ?
May 20 18:22:49 NOTICE[1224059824]: channel.c:1508
ast_set_read_format: Unable to find a path from G729A to ULAW
May 20 18:22:49 NOTICE[1224059824]: channel.c:1478
ast_set_write_format: Unable to find a path from ULAW to G729A
I was trying to replace the header. But looks like header contains some kind
of CRC
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Thomas Gallaway
> Sent: Tuesday, May 18, 2004 3:39 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk
What does it mean ?
May 2 20:37:21 WARNING[1205250992]:
chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386,
dropping
Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux
from
cvs checkout -r v1-0_stable asterisk
Hi Eric and Steve :)
My fax numbers (both Canon faxes)
3717201651
3726062501
- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, April 24, 2004 8:46 PM
Subject: Re: [Asterisk-Users] TxFax/SpanDSP problems
> On Sat, 2004-04-24 a
While calling to H323 peer
*CLI>
1:22:59.944 H225
Caller:81e5c48 assert.cxx(105)
PWLib Assertion fail: Invalid array element, file
/root/pwlib/include/ptlib/array.h, line 1183, Error=115
bort, ore dump, gnore?*CLI>
*CLI> show versionAsterisk CVS-04/22/04-23:56:0
I can provide low-priced termination service using *.
Write me offlist for more info.
- Original Message -
From: "Simon Brown" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 23, 2004 2:26 AM
Subject: [Asterisk-Users] SIP/IAX termination provider in NZ
I am looking for a
Hi Eric !
I have the same problem with Canon fax mashine as you have. I have wrote an
email to Steve (developer of spandsp) some weeks ago and got no answer how
to fix the problem :(
Looks like connection was dropped by fax mashine without any reason
- Original Message -
From: "Eric Wiel
t may speak with asterisk ! So, and last bug.. when I enable 2 codec in both version, I
need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set
codec by destination? ( like SIP )
I try use 2 cnannels at the same time, but asterisk
down with segmentation fault...
Thanks,Serge.
ation? ( like SIP )
I try use 2 cnannels at the same time, but asterisk down with
segmentation fault...
Thanks,Serge.
When calling from Zap (E100P) to ATA186 (SIP) *
hanged up...
below is 'show channels' command
output:
Channel (Context Extension Pri
) State Appl.
Data SIP/565-adc3
(voip
1 ) Up
AppDial (Outgoing
Line) Zap/31-1
(i
ation? ( like SIP )
I try use 2 cnannels at the same time, but asterisk down with
segmentation fault...
Thanks,
Serge.
--
Бесплатный почтовый ящик предоставлен http://webmail.delfi.lv
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http://lists.digiu
!
Serge
From: Anon <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Windows Drivers for Wildcard FXO Card
Date: Sun, 18 Apr 2004 06:39:33 -0600
On Friday 16 April 2004 09:37 am, Tony Mountifield wrote:
> In article <[EMA
t may speak with asterisk ! So, and last bug.. when I enable 2 codec in both version, I
need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set
codec by destination? ( like SIP )
I try use 2 cnannels at the same time, but asterisk
down with segmentation fault...
Thanks,Serge.
Hi Steve and all !
Unable to send fax from * to a fax machine...
File name is '/root/faxes/8400100-1081935748.808.tif'Changed from phase
0 to 2Slow carrier upSlow carrier downSlow carrier
up<<< NSF: 20 00 00 11 80 00 8a 49 10 43 53 43 20 54 45 4c 45 43 4f
4d 20 20 20 20 20 00 7b 00 80 8
* console just prints:
-- Hungup 'Zap/1-1
Thank you
Serge
From: James Golovich <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problem with Manager Originate
Date: Sun, 4 Apr 2004 15:02:42 -0400 (EDT)
On Sun, 4 Apr 2004, Serg
Hi
I am trying Manager interface for originate a call. This is what I get
---
Action: Originate
Exten: 555
CallerID: test <6656>
Context: local
Timeout: 600
Channel: SIP/8782
Priority: 1
Response: Error
Message: Originate failed
What do I do wrong?
Thank you
http interface. This can be
done using Tomcat. This means that your * installation has to have Tomcat
running as well. That might have security implications for your
installation.
Serge
From: John Todd <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subje
Hi!
I need your advice. My problem is that I have very
bad sound quality calling to cellular phone via asterisk router.There are
some kind of noises and echo effects when you try to speak louder.
I have the following components in my call routing
schema: - PBX with E1 port. - asterisk r
Is it possible to have ActionID on manager interface messages if call is
originated from a .call file?
From: Nicolas Bougues <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Identifying a call with manager interface
Date: Fri, 19 Mar 2004 16:17
Web Service is not available publicly. One of the reasons is that I do not
have a coumputer outsde of firewalls I could deploy it on.
Serge
From: "C. Johnson" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users]
.call file?
May be the same thing can be done for voice mail terminated call?
Serge
From: Greg Renouf <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: Asterisk_users_mail_list <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Web service to start a conference and
voicemail
Date:
. The same
thing might happen with fax or modem, but I have not seen that happening
yet.
Is there a way to identify when if there is a voice mail machine of the
line, fax or a modem?
Thank you
Serge
_
Get business advice and
Where i can download bluetooth support for * ?
How to send fax to analogue Fax device via PRI card (TE405P) from * ?
Should I use any kind of Fax emulation software or something like this?
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How to send fax to analogue device via PRI card (E100P) using * ?Should
I use any kind of emulation software or something like
this?
Thanks all,
I will install Linux. Your advice, what is better? RH 8 ? RH9 ? Mandrake?,
That was without problem.
I need converter SIP<>H.323 and H.323 codec converter.
Thanks.
Serge.
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL P
So, sorry I have general question , h.323 dont work on FreeBSD + asterisk
???,,, I need converter h.323 <> sip and codec converter for h.323.
I use FreeBSD 5.2.
>
Thanks all,
Serge.
- Original Message -
> From: "NetOne Administrator" <[EMAIL PROTECTED]
So, sorry I have general question , h.323 dont work on FreeBSD + asterisk
???,,, I need converter h.323 <> sip and codec converter for h.323.
I use FreeBSD 5.2.
Thanks all,
Serge.
- Original Message -
From: "NetOne Administrator" <[EMAIL PROTECTED]>
To: <[EMAIL
erisk on Feebsd , pls. HELP !
> On Monday 01 March 2004 13:02, William Waites wrote:
> > On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote:
> > > server dont have any sound device ( I think:) )
> > > Why noone make normal Makefile and FAQ for FreeBSD Asterisk..
D]>
Sent: Sunday, February 29, 2004 7:30 PM
Subject: Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
> On Sun, Feb 29, 2004 at 04:26:17AM +0100, Serge wrote:
> > Hello,
> >
> > Pls. help !
> > I have server on Freebsd 5.2 and don't may install a
nt: Sunday, February 29, 2004 7:30 PMSubject: Re:
[Asterisk-Users] Asterisk on Feebsd , pls. HELP !> On Sun, Feb
29, 2004 at 04:26:17AM +0100, Serge wrote:> > Hello,>
>> > Pls. help !> > I have server on Freebsd 5.2 and
don't may install asterisk , follo
Thanks William, it's get.
but new problem:
===
chan_oss.c:39:31: machine/soundcard.h: No such file or directory
chan_oss.c: In function `send_sound':
chan_oss.c:177: error: syntax error before "abi"
chan_oss.c:179: error: `SNDCTL_DSP_GETOSPACE' undeclared (first use
[db1-ast/libdb1.a] Error
2su-2.05b#---
have any idee?
Thanks,
Regards,
Serge.
interface.
How can I identify events that are related to the calls started via spool? I
tried to pass additional variables in the call using SetVar: statement, but
they do not get propagated into events.
Is there a way to do what I need?
Thank you
Serge
interface.
How can I identify events that are related to the calls started via spool? I
tried to pass additional variables in the call using SetVar: statement, but
they do not get propagated into events.
Is there a way to do what I need?
Thank you
Serge
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diling with OH323 will be appreciated.
Thank you,
Serge
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from SIP to H233 (OH323) or
from H323 to SIP
If I dial SIP to SIP on the same extention then *there is* ringing
indication.
I wander myself how to fix this problem.
Serge
From: Matt Lawson <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Use
created by
terminating side of the call)
Is there a way to make DISA application to generate ringing tone back to the
handset of the originating end point?
Thanks,
Serge
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Hi All
I am having this problem when setting up a H323 call.
Can anybody tell me what is going on?
Thanks
Serge
--
NOTICE[245776]: File chan_oh323.c, Line 1293 (oh323_write): H323:1637:
Format changed from 4 to 8.
WARNING[245776]: File frame.c, Line 76 (ast_smoother_feed
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