Hi all,
I have a PC working with a DIVA Eicon Server 4BRI during a lot
of time. Now I can't make call but I can receive calls.
I load diva with command: divactrl load -c 1 -f ETSI -u -t 0
Country: Spain
Isdnmode: point to point
My capi.conf is the next:
[global]
mode=immediate
?
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano Revuelto
Enviado el: miércoles, 19 de mayo de 2004 12:00
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] CAPI Eicon Diva Server 4BRI
Hi all,
I have a PC working
Hi,
I have a problem with a Eicon Diva Server 4 BRI. I have 4 BRI ISDN and
11 number for these 4 ISDN. At first I have connected one of these 4 ISDN.
When I try to call I receive the next trace:
-- Executing ChanIsAvail(SIP/716-b0cd,
We are going to do this test next week. I will say the result
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jim
Rosenberg
Enviado el: domingo, 29 de febrero de 2004 1:15
Para: Asterisk
Asunto: [Asterisk-Users] PCphoneline FXO to FXS
If your BG 101 is in intranet, try to adjust your qualify parameter to
60.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew B
Marlowe
Enviado el: viernes, 27 de febrero de 2004 2:08
Para: [EMAIL PROTECTED]
Asunto: RE:
Title: Mensaje
Hi
all,
For a few days we have a veryextrange
problem. We have an intranet with Budgetone and others SIP Phones.
In the
extranet We HaveBudgetone Phones. The whole system was working well
between the extranet and the intranet until a few days ago.
When
we try to speak
Title: Mensaje
Hola,
ahi va la sección [es] para el indications.conf
[es]
description = Spain
ringcadence = 1500,3000
dial = 425
busy = 425/200,0/200
ring = 425/1500,0/3000
congestion =
425/200,0/200,425/200,0/200,425/200,0/600
callwaiting =
425/175,0/175,425/175,0/3500
dialrecall =
Hi all,
I will ptobe your answers tomorrow. I'll say the results.
Thanks for all.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sascha
Knific
Enviado el: martes, 10 de febrero de 2004 22:08
Para: [EMAIL PROTECTED]
Asunto: AW: [Asterisk-Users]
Hi all,
anyone could help me with capi.conf?. I have installed an Eicon
Diva Server 4BRI. I have 2 EuroISDN BRI lines,
First line number: 951014943
Second line number: 951014944
I try to do 4 calls but, I can't do more than two call.
My capi.conf is the next:
You must use Monitor Application
Happy New Year,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edoardo
Borghesi [fabbricadigitale]
Enviado el: viernes, 02 de enero de 2004 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Call recording
PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Masakazu
Nakano
Enviado el: domingo, 21 de diciembre de 2003 5:37
Para: [EMAIL PROTECTED]
CC: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] RxFAX application
Hi sergio
On Fri, 19 Dec 2003 14:49:15 +0100
Sergio Serrano Revuelto [EMAIL PROTECTED
Hi all,
I have tested RxFAX application through X100P card. When Fax
arrive i obtain the next trace:
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetMusicOnHold(Zap/1-1, random) in new stack
-- Executing
Hey Srs.
I have a little problem with the next scenario:
Internal Phone(801)--Asterisk(public IP) --INTERNET--ADSL
Router--Budgetone(716)
|--ADSL Router--Budgetone(717)
My sip.conf is the next:
[general]
port = 5060 ; Port to bind to
bindaddr =
Next configuration must work:
zaptel.conf
fxoks=1-4
loadzone=fi
defaultzone=fi
Zapata.conf
[channels]
group=1
context=internt
signalling=fxo_ks
channel=1-4
Srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de JanM
Enviado el: jueves, 20 de
Title: Mensaje
try to
cvs
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Quan Le
TrungEnviado el: jueves, 13 de noviembre de 2003
10:43Para: [EMAIL PROTECTED]CC:
[EMAIL PROTECTED];
[EMAIL PROTECTED]Asunto:
Try to load module manually: modprobe wcfxo; ztcfg -
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Cameron
Palmer
Enviado el: viernes, 31 de octubre de 2003 6:27
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] 2 X100Ps give error
I
or address (6)
cameron.
On Fri, 31 Oct 2003, Sergio Serrano Revuelto wrote:
Try to load module manually: modprobe wcfxo; ztcfg -
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Cameron
Palmer Enviado el: viernes, 31 de
I need connect up to 100 analog phone to a H.323 network through *. I
think use TE410P, But I need to know what channel bank is better. I use
E1 lines
Any idea?
Thanks in advance,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de DUSTIN
WILDES
Title: Mensaje
AVM
Fritz it good for Asterisk. A little difficult to configure but not
impossible.
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tomica
CrnekEnviado el: jueves, 16 de octubre de 2003 14:36Para:
[EMAIL PROTECTED]Asunto:
Hi,
I have a problem with sip.conf. After some hours my sip
phone(netergy) hangs. In clonse appears the next logs repeatly:
10 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.0.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80
From: asterisk sip:[EMAIL
Title: Mensaje
Yes yo
can do it.
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
e-smithEnviado el: miércoles, 24 de septiembre de 2003
15:02Para: [EMAIL PROTECTED]Asunto:
[Asterisk-Users] Using Asterisk in an netted
scenario
Hi,
Just
exten=XXX,1,Dial(h323/3|17|tTm)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 11:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can
Could you send me your h323.conf and you gnugk.ini?
Sergio Serrano Revuelto
Responsable de Consultoría
Avanzada 7, S.L.
Teléfono / Fax: +34 951 01 49 47 / +34 951 01 09 22
www.avanzada7.com
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Try to add gwprefix in oh323.conf after your alias. You must know that
you can configure * gw in gnugk.ini or in oh323.conf. I recommend you
put in your oh323.conf.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el:
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 12:27
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?
Try to add gwprefix in oh323.conf after your alias
You can try AVM FRITZ with chan_capi from kapejod.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de YO Internet
Information
Enviado el: lunes, 22 de septiembre de 2003 0:03
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] ISDN BRI hardware
Hi all,
when I try register my netergy SIP Phone with *, I can't do it
due to the next message:
1 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a
From: asterisk sip:[EMAIL PROTECTED];tag=as34fa433f
To:
Title: Mensaje
Hi,
I
would like to configure a stage for SIP phones. This stage would be the
next:
two
netergy SIP phones connected to Asterisk through chan_sip.
one
X100P or AVM FRITZ to outside lines.
I
think that sip.conf would be the next:
;;
SIP Configuration for
I have the same problem,
Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0
Call-ID: [EMAIL PROTECTED]
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
To: 704sip:[EMAIL PROTECTED]
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.0.154:5060
Contact: sip:[EMAIL PROTECTED]:5060
Max-Forwards: 70
];tag=230b0-e0
instead of this:
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
Jan.
On 19-09 08:38, Sergio Serrano Revuelto wrote:
I have the same problem,
Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0
Call-ID: [EMAIL PROTECTED]
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
];tag=230b0-e0
instead of this:
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
Jan.
On 19-09 08:38, Sergio Serrano Revuelto wrote:
I have the same problem,
Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0
Call-ID: [EMAIL PROTECTED]
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
HI, I am probing chan_modem_i4l again with AVM FRITZ but I can hear
nothing in phone outside of asterisk, I explain
Phone 1-- AVM_FRITZ--Asterisk-- Phone 2
From Phone1 to Phone 2 I can hear, but
From phone 2 to phone 1 I can't hera nothing.
Any idea?
srsergio
-Mensaje original-
De:
Hi,
I would like to know how suppress number for outside dialling in
CDR table. For example, if I need press 9 key to make an outside call, I
would like that the number in dst field in cdr table was the outside
number without 9 key. It's possible?
Thanks in advance,
srsergio
Hi, I would like to know how do two things.
First, it is possible simulate PBX scenary?, I explain. I would like
that when an user press 9(outgoing key) asterisk will generate a new
dial tone in H.323 EP and then the user could press number for dial.
Second, it's possible modify time interdigit.
Hi, I would like to know how do two things.
First, it is possible simulate PBX scenary?, I explain. I would like
that when an user press 9(outgoing key) asterisk will generate a new
dial tone in H.323 EP and then the user could press number for dial.
Second, it's possible modify time interdigit.
In cdr table or in /var/log/asterisk/cdr-csv/Master.csv
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
[EMAIL PROTECTED]
Enviado el: miércoles, 16 de julio de 2003 23:54
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Dial SessionTime
Hi
Gsm is wav in 8/mono
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Scott
Stingel
Enviado el: lunes, 14 de julio de 2003 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] .gsm voice format
Hello-
What is the .gsm format? Ie: what's
Hi all,
I have probe cdr feature again and I realize when I make a call
from H.323 endpoint, I don't see any log in cdr table. My asterisk box
is the next:
AVM FRITZ-- |
|--EP
|- ASTERISK -OH323GATEKEEPER-
|--EP
]:
amaFlags=billing
Michael.
Sergio Serrano Revuelto wrote:
Hi all,
I have probe cdr feature again and I realize when I make a call
from
H.323 endpoint, I don't see any log in cdr table. My asterisk box is
the next:
AVM FRITZ-- |
|--EP
You must put and fax exten in your context:
For example:
[default]
...
exten = fax,1,Dial(Zap/2|30|d)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joe
Antkowiak
Enviado el: miércoles, 02 de julio de 2003 21:13
Para: [EMAIL PROTECTED]
www.avanzada7.com
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de nathan
Enviado el: miércoles, 28 de mayo de 2003 12:18
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] VOIP phone suppliers in the UK?
Hi All,
Can anyone recommend a supplier and/or a
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