i am set var Set(CDR(accountcode)=forkcdr-test) into agiphp
probe
$agi-exec('Set(CDR(accountcode)=5)');
$agi-exec('SetAccount','123123123');
and no work ...
how to solutions.
thanks people!
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i am install asterisk with asterisk-gui , the clients have spa841, but I am
call number outsite celular and I try to transfer the call to client sip,
not show XTRAN in the phone.
any idea??
configuration bad ???
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i am conecting to the provider voip with 50 logings and can not use trunk
the configuring sip.conf with
...
register = 10001:[EMAIL PROTECTED]
register = 10002:[EMAIL PROTECTED]
register = 10003:[EMAIL PROTECTED]
[troncal-1]
type=peer
secret=abc
username=10001
host=xxx.xxx.xxx.xxx
create nodes and links /proc/zap
On Jan 16, 2008 3:39 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
Make sure asterisk is in the dialout group in /etc/passwd
The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout,
and if you're using the gentoo ebuild of asterisk, it'll run
any version of asterisk not create nodes into /proc/zap
create to command, view into make file how to create nodes
On Jan 16, 2008 8:48 PM, Walter Willis [EMAIL PROTECTED] wrote:
create nodes and links /proc/zap
On Jan 16, 2008 3:39 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
Make sure
search , firewall, and confiration the software. the configuration the user
is bad ???
use asterisk2billing it is good
On 10/28/07, ram [EMAIL PROTECTED] wrote:
On 10/27/07, wassim darwish [EMAIL PROTECTED] wrote:
Hi:
Iam using an asterisk server with astcc ,iam facing a problem
view http.conf
bindaddr=0.0.0.0
On 10/10/07, Sanjoy Rath [EMAIL PROTECTED] wrote:
Hello,
When I click on User menu, I get loading screen status. It runs
indefinitely without showing me
the user list and the user admin menu.
Any thoughts ?
Thanks,
Sanjoy.
how to loging agent asterisk 1.4.13?
thanks
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HOw to call queues in asterisk ?
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i am configured asterisk-gui the Queue Extension Configuration but
configure and register into queue.conf :
[6]
fullname = Call Center
strategy = ringall
timeout = 5
wrapuptime = 5
autofill = yes
autopause = no
maxlen = 0
joinempty = no
leavewhenempty = no
reportholdtime = yes
musicclass =
i am have asterisk overnat with modem router dls-2640t or zyxel 660, the
clients it is over nat.
need cofiguration special ??
thanks .
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I install asterisk 14.10.1 and update to asterisk 1.4.11 but the terminals
when call to extensions no work rining but not work voice, the error :
[Aug 15 19:42:56] WARNING[2171] chan_sip.c: Maximum retries exceeded on
transmission [EMAIL PROTECTED] for seqno 101 (Critical
Response)
[Aug 15
thank you.
On 8/17/07, Steven [EMAIL PROTECTED] wrote:
Ahh, I see.
Good point.
--
--
Steven
http://www.glimasoutheast.org
Steve Totaro [EMAIL PROTECTED] wrote in message news:
[EMAIL PROTECTED]
Steven wrote:
I am curious.
Why would one need to do this?
If a phone
i have the problem in the hardware linksys pap2t, I am install asterisk with
asterisk-gui and work fine but the hangup the phone (linksys pap2 t), no
tone and sound like tu,tu, tu , tu , tu , tu ,
tuu
what is the problem with phone ???
add param
Thanks you.
work
On 8/16/07, Troy Ayers [EMAIL PROTECTED] wrote:
Walter Willis wrote:
i have the problem in the hardware linksys pap2t, I am install
asterisk with asterisk-gui and work fine but the hangup the phone
(linksys pap2 t), no tone and sound like tu,tu, tu , tu , tu , tu
hot to asterisk multiport...???
example 5060, 5061, 5080
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i am have x100p and not work fine, no detect polarity, and much problems
with asterisk 1.2 to up.
:S
On 7/25/07, Mike Wright [EMAIL PROTECTED] wrote:
John Novack wrote:
Mike Wright wrote:
Just purchased a Motorola Wildcard X100P ...
but the button pressed generates no tone; on button
install asterisk with x100p clone; the problem is that call me and hangup
but the interface zap not detect the hangup and the line open.
the error is :
The 'reload' command is deprecated and will be removed in a future release.
Please use 'module reload' instead.
-- Reloading module
i am have x100P clone, and install asterisk 1.4 and out call normaly and
hangup in xlite to zap but call to asterisk for zap channel nop pass to
xlite and the caller hangup the asterisk not detect.
what is the problem ???
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look my zapata.conf
[channels]
context=default
switchtype=national
signalling=fxs_ks
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
i am find manual asterisk with ldap. any url???
thanks
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how to recovery password spa3000?
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I have like clients several spa 3000, problem is that spa3000 is not
registered or something by the east style problem must to be by
bandwidth? spa3000 verifies bandwidth qeu can use and that is
registered or no?very I am intrigued with this problemilla. Thanks your help.
i am asterisk server with spa 3000 client in direfent site, the problem is what not regitser spa in deterninate hours,.the client becomes disconnected per moments, the register not show with command:sip show peers
not show iphey can explain to me like adjustment that?he is bandwidth???THANK!!
i am install server asterisk with 266Mhz , disk scsi and 380 Megas of ram.i am install into system the licence the codec g729 and the call it happens through the asterisk??the packets nessesarity through server asterisk ?
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look udev rules???On 7/23/06, Frank Darner [EMAIL PROTECTED] wrote:
Tom Lynn: perhaps not what you're looking for, but reading thru your config, it looks like you've mis-spelled 'echo cancel' as 'echo cancle'you are right, typothank you___
--Bandwidth
i am search, pre paid software for asterisk and exist several, as they
recommend to me?
mcc, as2bill, etc etc
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I have problem, install asterisk the cvs
cvs checkout -r v1-2-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
compile normal, install normal, user with modem motorola.
the cal sip to motorola work fine, but sip to sip no ring
___
--Bandwidth
log the call
-- Unregistered SIP '1234'
-- Registered SIP '1234' at 201.230.97.110 port 9906 expires 3600
-- Added extension '1234' priority 1 to sipregistrations
-- Saved useragent xlite release 3010n stamp 19039 for peer 1234
-- Executing Dial(SIP/1234-cfb0, SIP/[EMAIL
not work fine2005/1/25, Pawel Jaskorzynski @ Sokolka [EMAIL PROTECTED]:
Hello,could You spare some more details about this? Any source code modifications?Greetings,Pawel- Original Message -From: Jefferson Carvalho
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial
the openser add user not more.
using serweb send mail and other steps
2005/11/8, Krishna Sumanth Chava [EMAIL PROTECTED]:
Hi,
i am trying to set up SER and Asterisk... I am new to SER
and am having problems registering users to SER. i had the
Default user Admin registered to my softphone but am
the ser an asterisk run in the same box???
redirect host + port :)
2005/11/4, harry gaillac [EMAIL PROTECTED]:
Hello,I wish to setup this scheme:ser-0.9.4asterisk-1.2-bêtapolycom ip300 phonesasterisk 5050-- |firewall+nat|-192.168.ser 5060---My ip phones use ser as outbound sip proxy and
equivoco), es de que parte de EEUU quieres que sean los números
Entrantes (Para que puedan
llamarte a ti)
Saludos.
Juan.
-Mensaje original-
De:
[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]]En nombre de Walter Willis
Enviado el: Lunes, 31 de Octubre
de 2005 07:48 p.m
senecacom.net es tu dominio??
por cierto no das planes de reseller???
jejejeje.
mi ingles es
El día 1/11/05, Walter Willis [EMAIL PROTECTED] escribió:
las llamadas serian solo salientes, por lo que se, pero que quieran el
servicio de recepcion de llamadas, cuanto costaria todo eso???
podrias
thanks!!!
ask.
the fullcontact varchar(128) is added to
#
# Table structure for table `sip_buddies`
#
CREATE TABLE `sip_buddies` (
`id`int(11)NOTNULLauto_increment,
`name`varchar(80)NOTNULLdefault'',
`accountcode`varchar(20)defaultNULL,
`amaflags`varchar(7)defaultNULL,
tengo un asterisk, alguien conoce algun proveedor que brinde el
sistema de linkar mi asterisk a su servicio para tener tarifa plana a
eeuu.
para llamar por 4 conexiones al miamo tiempo desde mi asterisk?
me parece haber visto que se configuraba con una troncal iax2
2005/10/31, [EMAIL
.
I think that VoicePulse Connect might be able to do what you want. You
can call to the USA and I think that it does work with IAX2 Trunking.
http://connect.voicepulse.com/
-Rusty
On 10/31/05, Walter Willis
[EMAIL PROTECTED] wrote:
tengo un asterisk, alguien conoce algun proveedor que brinde el
Communications, LLC
[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Walter Willis
Sent: Monday, October 31, 2005
5:08 PM
To: Rusty Dekema; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
llamdas por 4 lineas a eeuu
2005/10/31, Walter Willis [EMAIL PROTECTED]:
Tengo un asterisk en al oficina de un cliente , que quiere hacer
llamadas ilimitadas a estados unidos; las llamadas tienen que ser al
mismo tiempo.
alguien ofrecio una conexion iax2 para los 4 usuarios.
no tienes algo mas atractivo se supone que
i am configure ser:
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
rewritehostport(192.168.0.183:5080);
};
};
an asterisk:
sip.conf
; config Xlite
[1234]
;context=sip
context=from-ser
type=friend
auth=md5
username=1234
secret=chooseapassword
;fromdomain=sorcier.com.pe ; para
I am tryting call client sip (SER) to client sip (Asterisk) and produce error:
Aug 21 18:48:13 WARNING[13370]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 18308
(Non-critical Response)
how to fix the problem
i am configure ser:
if
complemented the previous mail, cradled beginning the call in the
client sip (TO BE), gives congested tone me of, but the client
sip(asterisk) receives the call tone and comiensa to stamp, when I
answer I telephone shows hung up to me and the other continues
sounding equal. algun parametro forgets
i am install asterisk in gentoo linux,
#emerge zaptel
#emerge asterisk
#modprobe zaptel
#modprobe wcfxo
#asterisk -vvvc
localhost ~ # asterisk -vvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.8, Copyright (C)
+ o - va asi !!!
if (metho == INVITE ){
rewriteuriport(192.168.45.12: 5061)
forward(192.168.45.12,5061)
}
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i am server with ser(iptel) and asterisk.
the prosess is:
the xlite the connect to ser, the ser redirect to asterisk and call for x100p
after the two minutes the call, the call is cut.
the error is:
Asterisk Ready.
*CLI -- Executing Dial(SIP/sorcier.com.pe-41000490,
Zap/1/499732) in new
the idea is this:
I have a servant ASTERISK with two cards clone x100p, one of them I
want it to form so that it makes calls by the telefonica line and the
other to connect I telephone stops from ahi to be able to make calls.
the question is podra to become that?
for but information
I have a
the idea is this:
I have a servant ASTERISK with two cards clone x100p, one of them I
want it to form so that it makes calls by the telefonica line and the
other to connect I telephone stops from ahi to be able to make calls.
the question is podra to become that?
for but information
I have a
the asterisk suport NAT as ser?
or need modules from modules or special cofiguration?
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thank you, my English is terrible, I don't usually use
it and ti is not my language.
unfortunately they don't exist clever of mail in
another language. and I don't have with the one who to
practice it.
XD I go he is necessary to have to practice it but.
thanks you for the you help me.
yee
gracias por la ayuda
pero de todos modos tengo que practicarlo.
lo leo lo entiendo pero no lo escribo ni lo
habloo!
XD
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Visítanos
--- Walter Willis [EMAIL PROTECTED] escribió:
Fecha: Wed, 10 Nov 2004 11:15:40 -0600 (CST)
De: Walter Willis [EMAIL PROTECTED]
Asunto: the asterisk work with modem generic?
Para: [EMAIL PROTECTED]
i have modem generic
BCM v.90 56 kbps and modem 56k fax modem encore
(with
v.92
i have GTW V.92 Modem work with asterisk?
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___
Asterisk-Users
--- Walter Willis [EMAIL PROTECTED] escribió:
Fecha: Wed, 10 Nov 2004 12:58:02 -0600 (CST)
De: Walter Willis [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] calls go silent
Para: [EMAIL PROTECTED]
i have GTW V.92 Modem work with asterisk
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