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Grandstream Networks, nc.

GXP1400/1405 User Manual


Firmware version 1.0.1.73
Page 1 of 1
Last Updated: 07/2011
Grandstream Networks, nc.
GXP1400/1405 Small-Medium Business P Phone
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 2 of 35
Last Updated: 07/2011
TABLE OF CONTENTS
GXP1400/1405 USER MANUAL
BIENVENIDOS
!
INSTALLATION
4
E"UIPO DE
ENVASADO
4
CONEXI#N DEL
TEL$FONO 4
SEGU%IDAD
COMPLIANCES
4
&A%%ANT'
4
DESC%IPCI#N DEL
P%ODUCTO
5
USO DE LA
GXP1400/1405(
Fa)*l*ar*+arse ,-n la .an/alla
LCD (
%eal*+ar lla)a0as
/ele12n*,as
3
CONTESTA% LLAMADAS
TELEF#NICAS
11
Fun,*-nes 0el /el41-n- 0uran/e una lla)a0a
/ele12n*,a 1!
LLAMADA
FEATU%ES
15
MEDIDA DE PANTALLA LCD 6
XML 14
GU7A DE
CONFIGU%ACI#N
15
CONFIGU%ACI#N MEDIANTE
TECLADO 15
CONFIGU%ACI#N VIA NAVEGADO%
&EB 1(
C2)- 8uar0ar la ,-n1*8ura,*2n
C9ANGES5!
%e*n*,*- 0el TEL$FONO
%EMOTEL'5!
ACTUALI:ACI#N DEL SOFT&A%E '
CUSTOMI:ATION55
ACTUALI:ACI#N FI%M&A%E T%AV$S TFTP /
9TTP 55
DESCA%GA% A%C9IVO DE
CONFIGU%ACI#N 54
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
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Last Updated: 07/2011
%ESTAU%A% CONFIGU%ACI#N DE
F;B%ICA 55
7NDICE DE TABLAS
GXP1400 / 1405 MANUAL DEL USUA%IO
Ta<la 1= E>u*.-s 0e e)<ala?e 4
Ta<la != GXP1400 / 1405 C-ne,/-res
4
Ta<la 5= GXP1400 / 1405 Gu@a 0e
1un,*-nes 5
Ta<la 4= GXP1400 / 1405 Cara,/er@s/*,as .r*n,*.ales 0e un
V*s/a+- 5
Ta<la 5= GXP1400 / 1405 Es.e,*1*,a,*-nes 0el
Aar0Bare 5
Ta<la C= GXP1400 / 1405 Es.e,*1*,a,*-nes
/4,n*,as C
Ta<la D= De1*n*,*2n LCD D*s.la6
(
Ta<la (= LCD I,-n-s (
Ta<la 3= GXP1400 / 1405 <-/-nes 0el /e,la0-
3
Ta<la 10= GXP1400 / 1405 Fun,*-nes 0e
lla)a0as 14
Ta<la 11= MenE 0e C-n1*8ura,*2n 0e
Te,la0- 15
Ta<la 1!= Te,la0- Fl-B GUI 1C
Ta<la 15= C-n1*8ura,*2n 0el 0*s.-s*/*F- G
Es/a0- 13

Ta<la 14= C-n1*8ura,*2n 0el 0*s.-s*/*F- G A?us/es / A?us/es
<Hs*,-s 13
Ta<la 15= C-n1*8ura,*2n 0el 0*s.-s*/*F- G C-n1*8ura,*2n / C-n1*8ura,*2n
aFan+a0a !1
Ta<la 1C= C-n1*8ura,*2n 0e la ,uen/a
SIP !C
Ejemplos GU NTERFACE
GXP1400 / 1405 MANUAL DEL USUARO
http://www.grandstream.com/products/gxp_series/general/documents/gxp
21xx_gui.zip
1. Captura de pantalla de la configuracin de la pgina de entrada
2. Captura de pantalla de la pgina de estado
3. Captura de pantalla de la configuracin de Ajustes bsicos pgina
4. Captura de pantalla de la pgina de configuracin de usuario
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 4 of 35
Last Updated: 07/2011
avanzada
5. Captura de pantalla de la pgina de configuracin de cuenta SP
6. Captura de pantalla de configuracin Cambios Guardados Pgina
7. Captura de pantalla de la pgina Reboot
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 5 of 35
Last Updated: 07/2011
&el,-)e
GXP1400/1405 is a next generation small-to-medium business P phone that features 2 lines with 1 SP
account, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports
with integrated PoE (GXP1405 only), and 3-way conference. The GXP1400/1405 delivers superior HD
audio quality, rich and leading edge telephony features, personalized information and customizable
application service, automated provisioning for easy deployment, advanced security protection for
privacy, and broad interoperability with most 3
rd
party SP devices and leading SP/NGN/MS platforms. t
is a perfect choice for small-to-medium businesses looking for a high quality, feature rich P phone with
affordable cost.
Caution: Changes or modifications to this product not epress!" approved #" $randstream% or operation
of this product in an" wa" other than as detai!ed #" this User &anua!% cou!d void "our manufacturer
warrant".
Warning: '!ease do not use a different power adaptor with the $('1)00/1)0* as it ma" cause damage
to the products and void the manufacturer warrant".
Note:
+his document is su#,ect to change without
notice.
-eproduction or transmitta! of the entire or an" part% in an" form or #" an" means% e!ectronic or print%
for an" purpose without the epress written permission is not permitted.
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 6 of 35
Last Updated: 07/2011
Ins/alla/*-n
E"UIPMENT PACIAGING
Ta<le 1= E>u*.)en/ Pa,Ja8*n8
GXP1400/1405
Ma*n Case Yes
9an0se/ Yes
PA-ne C-r0 Yes
P-Ber A0a./-r Yes (GXP1400 only)
E/Aerne/ Ca<le Yes
Base S/an0 Yes
"u*,J S/ar/ Gu*0e Yes
CONNECTING 'OU% P9ONE
The connectors of the GXP1400/1405 are located on the bottom of the device.
Ta<le != GXP1400/1405 C-nne,/-rs
PC 10/100Mbps RJ-45 ports for PC (downlink) connection
LAN 10/100Mbps RJ-45 port for LAN (uplink) connection, integrated PoE (GXP1405 only)
P-Ber Ka,J 5V DC power port; UL Certified
9an0se/ Ka,J RJ9
9ea0se/ Ka,J RJ9
NORMAS DE SEGURDAD
El telfono GXP1400 / 1405 cumple con diversos estndares de seguridad de la FCC / CE y. El
adaptador de corriente GXP1400 / 1405 es compatible con el estndar UL. Utilice el adaptador
de corriente universal suministrado con el paquete GXP1400 1405 / solamente. La garanta del
fabricante no cubre daos provocados al telfono por adaptadores de corriente no compatibles.
GARANTA
Si usted compr su GXP1400 / 1405 de un distribuidor, por favor pngase en contacto con la
empresa donde compr el telfono por sustitucin, reparacin o reembolso. Si compr el
producto directamente de Grandstream, en contacto con las ventas Grandstream y
representante de servicio para un nmero de RMA (Return Materials Authorization) antes de
devolver el producto. Grandstream se reserva el derecho de corregir la poltica de garanta y sin
notificacin previa.
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 5 of 35
Last Updated: 07/2011
Fea/ures GXP1400/1405
LCD D*s.la6 128 x 40 pixel
Nu)<er -1 L*nes 2
Pr-8ra))a<le S-1/ Ie6s 3
EL/ens*-n M-0ule N/A
Fea/ures Bene1*/s
O.en S/an0ar0s
C-).a/*<*l*/6
SP RFC3261, TCP/P/UDP, RTP, HTTP/HTTPS, ARP/RARP, CMP,
DNS (A record, SRV and NAPTR), DHCP (both client and server),
PPPoE, TELNET, TFTP, NTP, STUN, SMPLE, SP over TLS, 802.1x,
TR-069
Su.er< Au0*- "ual*/6 Advanced Digital Signal Processing (DSP), Silence Suppression, VAD,
CNG, AGC
Ne/B-rJ In/er1a,es 10/100 Mbps Ethernet port, integrated PoE (GXP1405 only)
Fea/ure %*,A Traditional voice features including caller D, call waiting, hold, transfer,
forward, block, auto-dial, off-hook dial
A0Fan,e0 Fea/ures 2 line keys with dual-color LED and 1 SP account, 3 way conference,
graphic LCD, 3 XML programmable context sensitive soft keys, 5
navigation keys, 8 dedicated buttons for HOLD, TRANSFER,
CONFERENCE, VOLUME, HEADSET, MUTE/DND, SPEAKERPHONE,
SEND/REDAL
A0Fan,e0 Fun,/*-nal*/6 Customized downloadable ring-tones, SRTP, SP over TLS, multi-
language support and XML enabled, adjustable positioning angles, wall
mountable, AES encryption, automatic multimedia service (eg., weather
information)
GXP1400/1405
LAN In/er1a,e 10/100 Mbps Full/Half Duplex Ethernet port with auto detection
ntegrated PoE (GXP1405 only)
Gra.A*, LCD D*s.la6 128 x 40 pixel
EL.ans*-n M-0ule N/A
Call A..earan,e LED 2 Dual color (green/red) line keys
Pr-0u,/ OFerF*eB
Ta<le 5= GXP1400/1405 Fea/ure Gu*0e
Ta<le 4= GXP1400/1405 Ie6 Fea/ures *n a Glan,e
Ta<le 5= GXP1400/1405 9ar0Bare S.e,*1*,a/*-ns
L*nes 2 lines with 1 SP account, 3 XML programmable soft-keys
Pr-/-,-l Su..-r/ Support SP 2.0, TCP/UDP/P, PPPoE, RTP, SRTP by SDES, HTTP,
ARP/RARP, CMP, DNS, DHCP, NTP, TFTP, SMPLE/PRESENCE
protocols, TR-069, 802.1x
Support multiple SP accounts and up to 11 media channels
concurrently
Support SP PUBLSH method (RFC 3903), SP Presence package
(RFC 3856, 3863) for use of MFKs, SP Dialog package (RFC 4235)
Support for SP MESSAGE method (RFC 3428)
D*s.la6 Graphic LCD display, up to 4 level grayscale
Fea/ure Ie6s HOLD, TRANSFER, CONF, LNE 1, LNE 2, MSG, SPEAKERPHONE,
HANDSET, HEADSET, MUTE/DND, NAVGATON (5), VOLUME, 3
XML Programmable Soft keys
DeF*,e Mana8e)en/ NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed
devices including behind firewall/NAT
Auto/manual provisioning system, Web GU nterface
Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS,
DiffServ, MPLS)
Au0*- Fea/ures Full-duplex hands-free speakerphone
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723,1 (5.3/6.3K), G.729A/B, G.711 a/-law, G.726-
32, G.722 (wide-band), and iLBC codecs
n-band and out-of-band DTMF (in audio, RFC2833, SP NFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise
generation), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control (patent-pending) and packet delay and
loss concealment
HD audio handset with HD wideband audio codecs for excellent double-
talk performance
Tele.A-n6 Fea/ures ntuitive graphic user interface (GU), downloadable phone book (XML,
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
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Last Updated: 07/2011
Un*Fersal SB*/,A*n8 nput: 100-240VAC 50-60 Hz
P-Ber A0a./-r Output: +5VDC, 800mA, 4.0 W, UL
certified D*)ens*-n 186mm (W) x 210mm (L) x 81mm
(D)
&e*8A/
Unit weight: 0.7KG
Package weight: 1.1KG (GXP1400), 1.0KG (GXP1405)
Te).era/ure 32 -104

F/ 0 - 40

C
9u)*0*/6 10% - 90% (non-condensing)
C-).l*an,e FCC Part 15 (CFR 47) Class B
EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1
AS/NZS CSPR 22 Class B, AS/NZS CSPR 24, RoHS
UL 60950 (power adapter)
Ta<le C= GXP1400/1405 Te,An*,al S.e,*1*,a/*-ns
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 7 of 35
Last Updated: 07/2011
LDAP), support for anonymous call using privacy header, MLS (multi
language support)
Voice mail indicator, downloadable custom ring-tones, call hold, call
transfer (attended/blind), call forward, call waiting, caller D, mute, redial,
call log, caller D display or block, Do-Not-Disturb (DND) and volume
control
3-way conference, dial plan prefix, dial-plan support, off-hook auto dial,
auto answer, early dial and speed dial
Ne/B-rJ an0 Pr-F*s*-n*n8 Via keypad/LCD, Web browser, or secure (AES encrypted) central
configuration file, manual or dynamic host configuration protocol (DHCP)
network setup
Support NAT traversal using ETF STUN and Symmetric
RTP Support for EEE 802.1p/Q tagging (VLAN), Layer 3
ToS
F*r)Bare
U.8ra0es
Support firmware upgrade via TFTP or HTTP
Support for Authenticating configuration file before accepting changes
User specific URL for configuration file and firmware files
Mass provisioning using TR-069 or encrypted XML configuration file
A0Fan,e0 SerFer Fea/ures Message waiting indication, support DNS SRV Look up and SP Server
Fail Over, Support customizable idle screen via downloading XML by
HTTP/TFTP
Se,ur*/6 User and administrator level passwords, MD5 and MD5-sess based
authentication, AES based secure configuration file, SRTP, TLS,
802.1x media access control
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 8 of 35
Last Updated: 07/2011
Us*n8 /Ae GXP1400/1405
GETTING FAMILIA% &IT9 T9E LCD
GXP1400/1405 has a dynamic and customizable screen. The screen displays differently depending on
whether the phone is idle or in use (active screen).
Ta<le D= LCD D*s.la6 De1*n*/*-n
D*s.la6 I/e) De1*n*/*-ns
DATE AND TIME
Displays the current date and time. t can be synchronized with nternet time servers
LOGO NAME
Displays company logo name. This logo name can be customized via xml
screen customization. The maximum size for logo name is 22 characters in
English
NET&O%I
STATUS
Shows the status of network in the middle of the screen. t will indicate whether the network
is down or starting
STATUS BA%
Shows the status of the phone, using icons as shown in the next table
SOFTIE'S
The softkeys are context sensitive and will change depending on the status of the
phone. Typical functions assigned to soft-buttons are:
FO%&A%D ALL Unconditionally forwards the phone line to another phone
MISSED CALL This option shows unanswered calls to this phone.
NEXTSC% Press this button to toggle between idle screen, weather and P
Address.
%EDIAL Redials the last dialed-out number
END CALL Hangs up the call
Ta<le (= LCD I,-ns
I,-n LCD I,-n De1*n*/*-ns
SIP %e8*s/ra/*-n S/a/us I,-n=
S-l*0 connected to SP Server/P address received
SIP %e8*s/ra/*-n S/a/us I,-n=
BlanJ SP Proxy/Server not registered
9an0se/ S/a/us I,-n=
OFF - handset on-hook ON - handset off-hook
S.eaJer PA-ne S/a/us I,-n=
OFF - speakerphone off ON - speakerphone on
9ea0se/ S/a/us I,-n=
OFF - headset off ON - headset on
DND I,-n=
OFF - "Do Not Disturb disabled ON - "Do Not Disturb enabled
Calls F-rBar0e0 I,-n=
NDCATES all calls are forwarded. Please refer to the call forwarding procedures
MUTE I,-n=
NDCATES call is on MUTE during the call
S%TP I,-n=
NDCATES SRTP is enabled for the call
Ta<le 3= GXP1400/1405 IE'PAD BUTTONS
I,-n LCD I,-n De1*n*/*-ns
9OLD Place active call on hold
T%ANSFE% Transfer an active call to another number
CONF Press CONF button to connect Calling/Called party into conference
LINE 1 / LINE ! Switch between Line 1 and Line 2
Mute an active call; or use as DND button when the phone is in idle state.
Press HEADSET key to answer/hang up phone calls when using headset. t also allows
user to toggle between headset and speaker
Enable/Disable hands-free speaker
Enable/Disable handset mode; or used as SEND/REDAL
Press the four navigation keys to move up/down/left/right
Press the round button in the center to enter Keypad Configuration "MENU mode
when phone is idle. Or use it as ENTER key when in Keypad Configuration
Adjust volume by pressing " "or "+
0 G 3M NM O Standard phone keypad; press # key to send call; press * key to for VR functions
Realizar llamadas telefnicas
Auricular, los auriculares y el altavoz
El GXP1400 / 1405 le permite hacer llamadas telefnicas a travs del auricular, los auriculares o el
altavoz. Durante las llamadas activas, el usuario puede cambiar entre el auricular, los auriculares y el
altavoz pulsando las teclas correspondientes en el telfono.
Lneas duales con la cuenta SP
GXP1400 / 1405 puede soportar hasta dos lneas "virtualmente" asignadas a una cuenta SP. En
estado de descolgado, seleccionar una lnea libre y se escuchar el tono de marcacin. Para realizar
una llamada, seleccione la lnea que desea utilizar. El usuario puede cambiar las lneas antes de
marcar cualquier nmero pulsando el botn LNE.

Finalizacin de llamadas
Hay cuatro maneras de completar una llamada:
1. DAL: Para realizar una llamada telefnica.
Tome microtelfono descolgado
o presione el botn SPEAKER o presione el botn HEADSET
o pulse una tecla de lnea disponible para activar el altavoz
La lnea tendr un tono de marcacin
ntroduzca el nmero de telfono
Presione "#" o el botn TELFONO para enviar
2. REDAL: Para volver a marcar el ltimo nmero de telfono marcado.
Tome el auricular descolgado
o pulse el botn SPEAKER
o pulse una tecla de lnea disponible para activar el altavoz o en la pantalla de espera
Pulse la tecla variable REDAL
3. VA HSTORAL DE LLAMADAS: Para llamar a un nmero de telfono en la historia del telfono.
Pulse el botn MENU para abrir el men principal.
Seleccione Historial de llamadas y "Llamadas respondidas", entonces, "Llamadas perdidas" o
"Llamadas realizadas" o etc dependiendo de sus necesidades
Seleccionar el nmero de telfono con las teclas de flecha
Pulse Aceptar para seleccionar
Seleccione y pulse "Marcar" para marcar
4. VA AGENDA: Para llamar a un telfono desde la agenda del telfono.
r a la agenda pulsando la tecla de flecha ABAJO o presionando el botn de men y seleccionar
"Agenda"
Seleccione el nmero de telfono mediante las teclas de flecha
Pulse Aceptar para seleccionar
Seleccione y pulse "Marcar" para marcar
5. VA PGNA / NTERCOM: Servidor / PBX tiene que apoyar a la pgina / ntercom. Tambin,
GXP1400 / 1405 y PBX tienen que ser configurado correctamente.
Tome microtelfono descolgado
o presione el botn SPEAKER o presione el botn HEADSET
o pulse una tecla de lnea disponible para activar el altavoz
Pulse OK y la pantalla mostrar "Linex: PGNA"
Marque el nmero de pgina / ntercom
Pulse el botn "ENVAR" para marcar
Nota:
Tono y visualizacin del nmero marcado se produce despus de que el auricular est descolgado,
o se presiona el botn del auricular, o se presiona el botn de altavoz, o se selecciona la tecla de
lnea. Despus de marcar el nmero, el telfono espera 4 segundos (por defecto, sin clave de tiempo
de espera de entrada) antes de enviar y el inicio de la llamada. Pulse el botn "#" para anular los 4
segundos de retraso.

Realizacin de llamadas usando direcciones P
P Direct Call permite que dos telfonos para hablar unos con otros de una manera ad-hoc sin un
proxy SP. Las llamadas VoP se pueden hacer entre dos telfonos si:
Ambos telfonos tienen direcciones P pblicas, o
Ambos telfonos estn en una misma LAN / VPN con direcciones P privadas o pblicas, o
Ambos telfonos se pueden conectar a travs de un router con direcciones P pblicas o privadas
(con reenvo de puertos necesarios o DMZ)
Para realizar una llamada P directa, por favor, siga estos pasos:
Pulse el botn MENU para abrir MEN PRNCPAL
Seleccione "P Llamada Directa" utilizando las teclas de flecha
Pulse Aceptar para seleccionar
ntroduzca la direccin P de destino de 12 dgitos. (Por favor, consulte el ejemplo siguiente)
Presione la tecla OK para iniciar la llamada.
Por ejemplo: si la direccin P de destino es 192.168.1.60 y el puerto es 5062 (por ejemplo
192.168.1.60:5062), de entrada lo siguiente: 192 * 168 * 1 * 60 # 5062. La tecla "*" representa el
punto "."; la tecla "#" representa dos puntos ":". Pulse OK para marcar.
El GXP1400 / 1405 tambin es compatible con el modo P Llamada rpida. Esto permite que el
telfono para hacer llamadas P directas, utilizando slo los ltimos dgitos (ltimo octeto) de nmero
P del telfono de destino. Esto es posible slo si ambos telfonos estn en relacin con el mismo
LAN / VPN. Esto simula una funcin central utilizando el CMSA / CD sin un servidor SP. Se
recomienda el uso de P esttica controlada.
Para habilitar las llamadas P Quick, el telfono tiene que ser la primera configuracin. Esto se hace
a travs de la funcin WEB-setup. En la pgina "Configuracin avanzada", establecer el "uso rpido
modo de P-llamadas" a "S". Cuando se marca #xxx, donde x es 0-9 y xxx <= 255, una llamada P
directa a aaa.bbb. ccc.XXX se completa. "aaa.bbb.ccc" es a partir de la direccin P local,
independientemente de la mscara de subred. la #xx nmeros o #x tambin son vlidos. el 0 a la
izquierda no es necesaria (pero no est mal).
Por ejemplo:
192.168.0.2 192.168.0.3 llamando - marque # 3 seguido de #
192.168.0.2 llamando 192.168.0.23 - marque # 23 seguido de #
192.168.0.2 llamando 192.168.0.123 - marque # 123 seguido de #
192.168.0.2: marque # 3 y # 03 y # 003 resultados en la misma llamada - llamar 192.168.0.3
Nota:
Si tiene un servidor SP configurado, un P-P directa todava funciona. Si est utilizando STUN, el
Directo P-P
llamada tambin utilizar STUN. Configure el "uso de puertos aleatorios" a "No" al completar
llamadas P directas.
CONTESTAR LLAMADAS TELEFNCAS
Recepcin de llamadas
1. entrante Llamada individual: El telfono suena con el anillo de tono seleccionado. La lnea
correspondiente parpadea en rojo.
Respuesta de llamadas mediante la adopcin de auricular colgar o presionar ALTAVOZ o auricular o
presionando el botn de la cuenta de lnea correspondiente.

2. mltiples llamadas entrantes: Cuando otra llamada mientras tiene una llamada activa, el telfono
producir un tono de llamada en espera (tono entrecortado). Conteste la llamada entrante pulsando
su botn de lnea correspondiente. La llamada activa se pone en espera.
No molestar
1. Pulse el botn MENU y desplcese hacia abajo a "Preferencias".
2. Seleccione "No molestar" pulsando el botn men.
3. Utilice las teclas de flecha para desactivar o activar de "No Molestar".
4. Cuando est activado, habr un especial de "No molestar" icono que aparece en la pantalla. Esto
enviar la llamada entrante directamente al correo de voz.
Funciones del telfono durante una llamada telefnica
Llamada en espera / retencin de llamada
1. Hold: Poner una llamada en 'mantener' pulsando el botn "HOLD".
2. Reanudar: Retoma la llamada pulsando la LNEA parpadeando correspondiente.
3. Llamadas Mltiples: Coloca automticamente llamada activa en "USAR" al seleccionar otra lnea
disponible para realizar o recibir otra llamada. Tono de llamada en espera (tono entrecortado) audible
cuando la lnea est en uso.
Mute
1. Pulse el botn MUTE para activar / desactivar silenciar el micrfono.
2. El "ndicador de estado de lnea" mostrar "Linex: HABLAR" o "Linex: MUTE" para indicar si el
micrfono est silenciado.
Transferencia de llamada
GXP1400 / 1405 admite la transferencia de Ciegos y asisti:
1. Transferencia ciega: Pulse el botn "TRANSFER", luego marque el nmero y pulse el botn # para
completar la transferencia de la llamada activa.
2. Transferencia atendida: Pulse el botn "Linex" para hacer una llamada y automticamente colocar
la lnea activa en espera. Una vez establecida la llamada, teclee "TRANSFER" presione entonces el
botn ON LNE de la lnea de espera para transferir la llamada. Cuelgue el telfono mvil despus se
transfiere la llamada.
Nota:
Para transferir llamadas entre dominios SP, proveedores de servicios SP deben apoyar la
transferencia a travs de SP
dominios.
3 vas de conferencia
GXP1400 / 1405 puede albergar conferencias telefnicas y soporta hasta llamadas de conferencia de
3 vas.
1. niciar una llamada de conferencia:
Establecer una conexin con dos partidos

Pulse el botn CONF
Seleccione la lnea que desea unirse a la conferencia pulsando el botn de lnea correspondiente
2. Cancelar Conferencia:
Si despus de pulsar el botn "CONF", un usuario decide no tener una conferencia cualquiera,
presione HOLD
o el botn formacin original
Esta se reanudar una conversacin bidireccional
3. Conferencia Final:
Pulse RETENER para finalizar la llamada en conferencia y poner todas las partes en espera
Para hablar con una tercera parte individual, seleccione la tecla ON LNE correspondiente
Nota:
La parte que inicia la conferencia tiene que permanecer en la conferencia para toda su duracin, se
puede poner el partido en silencio, pero si no permanece en la conversacin. Adems, esto no es
aplicable cuando la funcin "Transferencia de llamada colgar" est encendido.
Mensajes de Voz (ndicador de mensaje en espera)
A MW roja parpadeante (ndicador de mensaje en espera) indica un mensaje en espera. Marque en
la casilla de correo de voz para recuperar el mensaje. Un VR pedir al usuario a travs del proceso
de recuperacin de mensajes.
Llame compartido Apariencia (SCA)
Los GXP1400 / 1405 brinda el telfono comparten lnea de llamada por norma Broadsoft. Esta
caracterstica permite a los miembros del grupo SCA a lneas SP compartidos y proporciona
supervisin del estado (inactivo, activo, progresando, espera) de la lnea compartida. Cuando hay
una llamada entrante designado para el grupo SCA, todos los miembros del grupo sern notificados
de una llamada entrante y ser capaz de responder a la llamada desde el telfono con la extensin
SCA registrado.
Todos los usuarios que pertenecen al mismo grupo SCA sern notificados por el indicador visual
cuando un usuario se apodera de la lnea y realiza una llamada saliente, y todos los usuarios de este
grupo no ser capaz de apoderarse de la lnea hasta que la lnea se remonta a una estado de reposo
o cuando la llamada se pone en espera. (Con la excepcin de cuando mltiples lneas de llamada
estn habilitadas en el lado del servidor).
En medio de la conversacin, hay dos tipos de retencin: Hold Hold Pblica y Privada. Cuando un
miembro del grupo pone la llamada en espera pblica, los dems usuarios del grupo SCA sern
notificados de esta por el botn rojo-parpadeo y ellos sern capaces de retomar la llamada desde su
telfono pulsando el botn de lnea. Sin embargo, si esta llamada se pone en privado-hold, ningn
otro miembro del grupo SCA podr reanudar la llamada.
Para habilitar la lnea de llamada compartida, el usuario tendra que registrar la cuenta de lnea
compartida en el telfono. Adems, tendran que vaya a "Configuracin" -> "Configuracin bsica" en
la interfaz de usuario web y establecer la lnea de "lnea compartida". Si el usuario requiere lneas de
llamada ms compartidas, el usuario puede configurar mltiples botones de lnea para ser botones
de "lnea comn" asociados a la cuenta.
Funciones de llamada
El GXP1400 / 1405 es compatible con las caractersticas tradicionales y avanzados de telefona
incluyendo identificador de llamadas, identificador de llamadas
w / nombre, desvo de llamada / transferencia / parque / retencin, as como del interfono /
localizador.
Ta<le 10= GXP1400/1405 Call Fea/ures
Ie6 Call Fea/ures
N50 Bl-,J Caller ID (for all subsequent calls)
Offhook and dial "*30.
N51 Sen0 Caller ID (for all subsequent calls)
Offhook and dial "*31.
NCD Bl-,J Caller ID (per call)
Offhook, dial "*67 and then enter the number to dial out.
N(! Sen0 Caller ID (per call)
Offhook, dial "*82 and then enter the number to dial out.
ND0 D*sa<le Call &a*/*n8 (per Call)
Offhook, dial "*70 and then enter the number to dial out.
ND1 Ena<le Call &a*/*n8 (per Call)
Offhook, dial "*71 and then enter the number to dial out.
ND! Un,-n0*/*-nal Call F-rBar0
Offhook, dial "*72. Then enter the number to forward the call and press "# or OK softkey.
ND5 Can,el Un,-n0*/*-nal Call F-rBar0
Offhook, dial "*73 and the phone will hang up.
N30 Bus6 Call F-rBar0
Offhook, dial "*90. Then enter the number to forward the call and press "# or OK softkey.
N31 Can,el Bus6 Call F-rBar0
Offhook, dial "*91 and the phone will hang up.
N3! Dela6e0 Call F-rBar0
Offhook, dial "*92. Then enter the number to forward the call and press "# or OK softkey.
N35 Can,el Dela6e0 Call F-rBar0
Offhook, dial "*93 and the phone will hang up.
MEDDA DE PANTALLA LCD y XML
Soporte telefnico P GXP1400 / 1405 tanto para aplicaciones simples y avanzadas XML: 1) Pantalla Custom
XML, 2) XML descargables Agenda. Para obtener ms informacin sobre cmo crear una agenda XML
descargable, creando una pantalla personalizada de inactividad y / o reprogramar las teclas variables en
GXP1400 / 1405, por favor visite nuestro sitio web en http://www.grandstream.com/support.

Gua de configuracin
I/e)
Call 9*s/-r6
S/a/us
PA-ne B--J
LDAP D*re,/-r6
Ins/an/ Messa8es
D*re,/ IP Call
Pre1eren,e
El GXP1400 / 1405 se puede configurar de dos maneras. En primer lugar, mediante el men de Configuracin
de Teclado en el telfono;
en segundo lugar, a travs del men de configuracin web integrada.
CONFGURACN MEDANTE TECLADO
Para acceder al men, pulse el botn redondo. Navegar por el men utilizando las teclas de direccin: arriba /
abajo e izquierda / derecha. Pulse el botn Aceptar para confirmar la seleccin de mens. El telfono sale
automticamente del modo MENU con una llamada entrante, el telfono est descolgado o el modo MENU es
dejado inactivo durante 20 segundos.
Pulse el botn MENU para entrar en la tecla Men Pad Key. Las opciones de men disponibles se muestran en
el cuadro 11.
Ta<le 11= Ie6 Pa0 C-n1*8ura/*-n Menu
Des,r*./*-n
Displays histories of answered, dialed, missed, and transferred and forwarded
calls
Displays the network status, account status, software version, MAC-address and
hardware version of the phone
Displays the phonebook and downloads phonebook XML
Displays the LDAP directory and downloads directory
Goes to instant messages
Dials P address for direct P call
Press &enu button to enter this sub menu including:
D- NOT D*s/ur<
DND (Do Not Disturb) function could be turned on or off in the "Do Not
Disturb menu.
%*n8 T-ne
Choose different ring tones in the "Ring Tone menu.
%*n8 V-lu)e
Press &enu button to hear the selected ring volume, press PQ or Q Q
to hear and adjust the ring tone volume.
LCD C-n/ras/
Press PQ or Q Q to adjust the LCD contrast.
D-Bnl-a0 SC% XML
The phone will download the custom idle screen if available.
Erase Cus/-) SC%
Custom idle screen will be erased and will be replaced with default
logo.
D*s.la6 Lan8ua8e
You can choose English, Simplified Chinese, Traditional Chinese,
Korean, Japanese, talian, Spanish, French, German, Portuguese,
Russian, Croatian, Hungarian, Polish, Slovenian which are built in the
phone. Users could select Automatic for local language based on P
location if available. Also, the phone will download secondary language
if available.
T*)e Se//*n8s
Press &enu button to choose the menu item
Press '' or follow the soft keys to return to the main menu
C-n1*8 Press &enu button to display the configuration selections:
SIP
To change SP server settings for SP account.
U.8ra0e
n this menu setting regarding the firmware server and Config server
can be changed. t also enables the user to make the phone attempt to
download new firmware.
Fa,/-r6 %ese/
Key in the physical/MAC address on back of the phone.
Press &enu button to reset FACTORY DEFAULT setting. Do not use
Factory Reset unless you want to restore factory settings.
La6er ! "-S
Configure 802.1Q/VLAN Tag and priority value.
Fa,/-r6 Fun,/*-ns
Press &enu to display the factory function items including
Au0*- L--.<a,J
Speak into the handset. f you hear your voice in the handset, your audio
works fine. Press &enu button to exit the mode
D*a8n-s/*, M-0e
All LEDs will light up.
Press any key on the keypad, to display the button name in the LCD.
Lift and put back the handset or press &enu button to exit the diagnostic
mode.
Press '' to return the main menu
Ne/B-rJ To enable/disable DHCPR to setup P-address, Net mask and Gateway address
%e<--/ Press &enu button to reboot the device
EL*/ Exit from this menu
Ta<le 1!= Ie6.a0 GUI Fl-B
Call 9*s/-r6
Call 9*s/-r6 I/e)s
MENU
Call 9*s/-r6
S/a/us
PA-ne B--J
LDAP D*re,/-r6
Ins/an/
Messa8e
D*re,/ IP Call
Pre1eren,e
C-n1*8
Fa,/-r6
Fun,/*-ns
Ne/B-rJ
%e<--/
EL*/
AnsBere0 Calls
D*ale0 Calls
M*sse0 Calls
Trans1erre0 Calls
F-rBar0e0 Calls
Ba,J
PA-ne B--J
NeB En/r6
D-Bnl-a0 PA-ne<--J XML
Ba,J
LDAP D*re,/-r6
V*eB D*re,/-r6
D-Bnl-a0 D*re,/-r6
Sear,A C-n1*8ura/*-n
Ba,J
Ins/an/ Messa8e
Clear All
Ba,J
Pre1eren,e
D- N-/ D*s/ur<
%*n8 T-ne
LCD Br*8A/ness
D-Bnl-a0 SC% XML
Erase Cus/-) SC%
D*s.la6 Lan8ua8e
Ba,J
C-n1*8
SIP U.8ra0e
Fa,/-r6 %ese/
La6er ! "-S
Ba,J
Fa,/-r6 Fun,/*-n
Au0*- L--.<a,J
D*a8n-s/*, M-0e
Ba,J
Ne/B-rJ
IP Se//*n8
PPP-E Se//*n8s
IP
Ne/)asJ
Ga/eBa6
DNS SerFer 1
DNS SerFer !
Ba,J
Ba,J
Clear All
NeB En/r6
Na)e=
Nu)<er=
A,,/=
C-n1*r) A00=
Can,el an0 %e/urn=
Sear,A C-n1*8ura/*-n
Sele,/ F*l/er
F*l/er Value
Ba,J
D- N-/ D*s/ur<
Ena<le DND
D*sa<le DND
Ba,J
%*n8 T-ne
De1aul/ %*n8
%*n81
%*n8!
%*n8 5
Ba,J
D*s.la6 Lan8ua8e
En8l*sA
CA*nese
Fren,A
S.an*sA
Ger)an
I/al*an
S
Se,-n0ar6 Lan8ua8e
Lan8ua8e F*le P-s/1*L
Ba,J
SIP
A,,-un/
SIP Pr-L6
Ou/<-un0
Pr-L6
SIP User ID
SIP Au/A ID
SIP PassB-r0
SIP Trans.-r/
Au0*-
SaFe
U.8ra0e
F*r)Bare
SerFer
C-n1*8 SerFer
U.8ra0e V*a
La6er ! "-S
(0!1"/VLAN Ta8
Pr*-r*/6 Falue
%ese/ Vlan C-n1*8
Ba,J
CONFGURACN VA NAVEGADOR WEB
El servidor Web incorporado GXP1400 / 1405 responde a las solicitudes HTTP / HTTPS GET / POST.
Pginas HTML embebidos permiten al usuario configurar el telfono P a travs de un navegador web
como E de Microsoft, Mozilla Firefox y Google Chrome.
Acceder al men de configuracin Web
Para acceder al men de configuracin Web del telfono
Conecte el ordenador a la misma red que el telfono1
Asegrese de que el telfono est encendido y muestra su direccin P
nicie un navegador Web en el equipo
ntroduzca la direccin P del telfono en la barra de direcciones del browser2
ntroduzca la contrasea del administrador para acceder a la configuracin Men3 Web
1 El ordenador habilitado para la Web tiene que estar conectado a la misma subred que el telfono. Esto
se puede hacer fcilmente mediante la conexin de la computadora al mismo hub o switch como el
telfono est conectado. En ausencia de un hub / switch (o puertos libres en el hub / switch), conecte el
ordenador directamente al telfono mediante el puerto de la PC en el telfono.
2 Si el telfono est conectado a una conexin activa a nternet, el telfono mostrar su direccin P. Esta
direccin tiene el formato: xxx.xxx.xxx.xxx, donde xxx es un nmero de 0 a 255 Necesitar este nmero
para acceder al men de configuracin Web. Por ejemplo, si el telfono muestra 192.168.0.60, por favor,
utilice "http://192.168.0.60" en la barra de direcciones de su navegador.
3 La contrasea predeterminada del administrador es "admin"; la contrasea del usuario final por defecto
es "123".
Nota:
Al cambiar la configuracin, siempre someterlos al pulsar el botn "Actualizar" en la parte inferior de la
pgina. Si, despus de haber presentado algunos cambios, ms ajustes se tienen que cambiar, pulse la
opcin de men necesarias.
Definiciones
Esta seccin describe las opciones de la interfaz de usuario de configuracin Web. Como se ha
mencionado, un usuario puede iniciar sesin como administrador o usuario final.
Funciones disponibles para el usuario final son:
Estado: Muestra el estado de la red, Estado de Cuenta, la versin del software y la direccin MAC del
telfono, y el estado del servicio.
Configuracin bsica: preferencias bsicas tales como fecha y hora, llaves multiusos y LCD
ajustes se pueden definir aqu.
Funciones adicionales disponibles para los administradores son:
Ajustes avanzados: para fijar la configuracin avanzada de red, configuracin del codec y los ajustes de
configuracin XML y etc
Cuenta: Para configurar la cuenta SP.
Ta<le 15= DeF*,e C-n1*8ura/*-n G S/a/us
MAC A00ress The device D, in HEXADECMAL format.
IP A00ress This field shows P address of GXP1400/1405.
Pr-0u,/ M-0el This field contains the product model information.
Par/ Nu)<er This field contains the product part number.
S-1/Bare Vers*-n Pr-8ra): This is the main firmware release number, which is always used for
identifying the software (or firmware) system of the phone.
B--/: Booting code version number
C-re: Core code version number
Base: Base code version number
DSP: DSP code version number
AuL: Aux code version number
S6s/e) U. T*)e This field shows system up time since the last reboot.
S6s/e) T*)e This field shows the current time on the phone system.
%e8*s/ere0 ndicates whether accounts are registered to the related SP server.
PPP-E L*nJ U. ndicates whether the PPPoE connection is enabled (connected to a modem).
SerF*,e S/a/us
GUI: shows the GU status: running or stopped
PA-ne: shows the phone status: running or stopped
C-re Du). Download core dump file for troubleshooting when necessary.
Ta<le 14= DeF*,e C-n1*8ura/*-n T Se//*n8s/Bas*, Se//*n8s
En0 User PassB-r0 This contains the password to access the Web Configuration Menu. This field is c a s e
sensitive with a maximum length of 25 characters.
IP A00ress The GXP1400/1405 operates in three modes:
1. D9CP )-0e: The GXP1400/1405 acquires its P address from the first
DHCP server it discovers on its LAN. The DHCP option is reserved for NAT
router mode. n DHCP mode, all the field values for the Static P mode are
not used (even though they are still saved in the Flash memory).
2. PPP-E )-0e: To use the PPPoE feature, set the PPPoE account settings
(PPPoE account D, PPPoE password and PPPoE service name). The
GXP1400/1405 establishes a PPPoE session if any of the PPPoE fields is
set.
3. S/a/*, IP )-0e: Configure all of the following fields: P address, Subnet
Mask, Gateway, DNS Server 1, DNS Server 2 and Preferred DNS Server.
(0!1L M-0e This option allows the user to enable/disable 802.1x mode on the phone. The
default value is disabled. To enable 802.1x mode, this field should be set to EAP-
MD5.
Once enabled, the user would be required to enter the following information below
to be authenticated on the network:
dentity
MD5 Password
L*ne Ie6s L This allows the user to configure the account mapped to each line key, as well
as enabling SCA (Shared Call Appearance) for the line.
Options available for Key Mode are :
1. Line
2. Shared Line
T*)e :-ne This parameter controls the date/time display according to the specified time zone.
f "Allow DHCP Option 2 to override Time Zone setting is checked, the time zone
will be overridden by the DHCP server.
Sel1GDe1*ne0 T*)e
:-ne
This parameter allows the users to define their own time zone.
The syntax is: s/0 -11se/ 0s/ U-11se/VM s/ar/ U//*)eVM en0
U//*)eV Default is set to: MT:WCMDTW5MM410MM1110
MT:WCMDTW5M
This indicates a time zone with 6 hours offset with 1 hour ahead which is U.S central
time. f it is positive (+) if the local time zone is west of the Prime Meridian (A.K.A:
nternational or Greenwich Meridian) and negative (-) if it is east.
M410MM1110
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3
rd
Tuesday.)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, . ,Sat)
Therefore, this example is the DST which starts from the first Sunday of April to
the
1st Sunday of November.
&ea/Aer U.0a/e By default, "Enable Weather Update: is set to "Yes. f set to "No,
weather information will not display on the phone.
Settings to customize the display of weather via:
City Code Enter city code
Update nterval Refresh time in minutes
Degree Unit Select Automatic, Fahrenheit or Celsius
This is displayed when "Enable Weather Update is set to "Yes and pressing the
'SwitchSCR' soft-key once.
LCD C-n/ras/ Set LCD contrast. Range from 0 to 20.
T*)e D*s.la6 F-r)a/ LCD time display in 12 hour or 24 hour format.
D*sa<le *nG,all DTMF
0*s.la6
Default is "No. This field is used to hide the keypad input during a call.
9EADSET Ie6 M-0e Default Mode:
- Toggle to Headset when using Speaker/Handset
- Dial, pick up call or hang up call using Headset
Toggle Headset/Speaker:
- toggle between using Headset and using Speaker
9ea0se/ TX 8a*n X0BY Set headset TX gain to -6, 0 or +6. Default is 0 db.
9ea0se/ %X 8a*n X0BY Set headset RX gain to -6, 0 or +6. Default is 0 db.
Ta<le 15= DeF*,e C-n1*8ura/*-n T Se//*n8s /A0Fan,e0 Se//*n8s
A0)*n
PassB-r0
Administrator password. Only the administrator can access the "Advanced
Settings and "Account Settings page. Password field is purposely blank for
security reasons after clicking update and saved. The maximum password length is
25 characters.
La6er 5 "-S This field defines the layer 3 QoS parameter. t is the value used for P Precedence
or Diff-Serv or MPLS. Default value is 12.
La6er ! "-S This contains the value used for layer 2 802.1Q/VLAN tag and 802.1p priority value.
Default setting is 0.
L-,al %TP .-r/ This parameter defines the local RTP port pair used to listen and transmit. t is the
base RTP port for channel 0. When configured, channel 0 will use this port _value
for RTP; channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024
to 65400 and must be even. The default value is 5004.
Use %an0-) P-r/ This parameter, when set to "Yes, will force random generation of both the local
SP and RTP ports. This is usually necessary when multiple GXPs are behind the
same NAT. Default is "No.
Iee.Gal*Fe *n/erFal This parameter specifies how often the GXP1400/1405 sends a blank UDP packet
to the SP server in order to keep the "hole on the NAT open. Default is 20
seconds.
Use NAT IP NAT P address used in SP/SDP message. Default is blank.
STUN SerFer P address or Domain name of the STUN server. STUN resolution result will display
in the STATUS page of the Web U.
F*r)Bare U.8ra0e an0
Pr-F*s*-n*n8
Allows the user to select the following options for firmware upgrade:
Always Check for New Firmware
Check New Firmware only when F/W pre/suffix changes
Always Skip the Firmware Check.
Firmware upgrade may take up to 10 minutes depending on network environment.
Do not interrupt the firmware upgrading process.
N-/e= Grandstream strongly recommends that the user upgrade firmware locally
in a LAN environment if using TFTP to upgrade. Please DO NOT interrupt the
upgrade process (especially the power supply) as this will damage the device.
XML C-n1*8 F*le
PassB-r0
The password used for encrypting the XML configuration file using OpenSSL. This
is required for the phone to decrypt the encrypted XML configuration file.
9TTP/9TTPS User Na)e The user name for the HTTP/HTTPS server.
9TTP/9TTPS PassB-r0 The password for the HTTP/HTTPS server.
U.8ra0e V*a This field allows the user to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
F*r)Bare SerFer Pa/A Defines the server path for the firmware server. t can be different from the
Configuration server which is used for provisioning.
C-n1*8 SerFer Pa/A Defines the server path for provisioning; it can be different from the firmware server.
F*r)Bare F*le
Pre1*L/P-s/1*L
Default is blank. f configured, GXP1400/1405 will request the firmware file with the
prefix/postfix and only the firmware with the matching encrypted prefix will be
downloaded and flashed into the phone.
This setting is useful for TSPs. End user should keep it blank.
C-n1*8 F*le
Pre1*L/P-s/1*L
Default is blank. f configured, GXP1400/1405 will request the config file with the
prefix/postfix and only the file with the matching encrypted prefix will be downloaded
and flashed into the phone.
This setting is useful for TSPs. End user should keep it blank.
All-B D9CP O./*-n 45
an0 O./*-n CC /-
-Ferr*0e serFer
Default is "Yes. This allows device gets provisioned from the server automatically.
Au/-)a/*, U.8ra0e This function is used by TSP. End user should NOT touch these parameters.
Default is "No. Choose "Yes to enable automatic HTTP upgrade and provisioning.
n "Check for upgrade every field, enter the number of minutes to check the HTTP
server for firmware upgrade or configuration changes. When set to "No, the phone
will only perform HTTP upgrade and configuration check once at boot up.
Au/Aen/*,a/e C-n1 F*le Default is "No. f set to "Yes, configuration file would be authenticated before
acceptance. End user should use default setting.
Ena<le T%G0C3 Default is "No.
ACS U%L URL for TR-069 Auto Configuration Servers (ACS).
T%G0C3 Userna)e Enter username for TR-069.
T%G0C3 PassB-r0 Enter password for TR-069.
SaFe Cre0en/*als Save TR-069 credentials. Default is "No.
Au/- L-8*n Auto Login TR-069 account. Default is "No.
Per*-0*, In1-r) Ena<le Enable periodic inform. Default is "No.
Per*-0*, In1-r) In/erFal When enabling periodic inform, set up the periodic inform interval.
Grandstream Networks, nc. GXP1400/1405 User Manual Page 22 of 35
Firmware version: 1.0.1.73 Last Updated: 07/2011
C-nne,/*-n %e>ues/
Userna)e
Enter the connection request username.
C-nne,/*-n %e>ues/
PassB-r0
Enter the connection request password.
Au/Aen/*,a/*-n Me/A-0 Select the authentication method among "No authentication, "Basic or Digest.
C-nne,/*-n %e>ues/
P-r/
Enter the connection request port.
PA-ne<--J XML
D-Bnl-a0
Selects the file download mode for the download server. Users can choose from
TFTP/HTTP/No.
PA-ne<--J XML SerFer
Pa/A
The URL/P address of the phonebook download server.
PA-ne<--J D-Bnl-a0
In/erFal
The interval at which the phonebook will be downloaded from the download server
(in Minutes). The default setting is 0.
%e)-Fe Manuall6Ge0*/e0
en/r*es -n D-Bnl-a0s
f set to "Yes, the phone will remove the manually-edited entries in the old
phonebook list before downloading the new file. The default setting is set to "Yes.
LDAP D*re,/-r6 P address or domain name of LDAP script server.
I0le S,reen XML
D-Bnl-a0
Enable XML dle Screen download via TFTP or HTTP. Select whether to "Use
Custom Filename or not, and define the "XML server path.
D-Bnl-a0 S,reen XML
A/ B--/Gu.=
The phone will download the idle screen xml file if set to "Yes. The default setting
is "No.
Use Cus/-) F*lena)e The phone will use custom filename specified in XML server path if set to "Yes.
The default setting is "No.
I0le S,reen XML SerFer
Pa/A=
Specify the idle screen XML server path.
XML A..l*,a/*-n Enter server path for XML application.
S-1/Je6 La<el Defines the softkey label for the XML application.
O11A--J Au/- D*al To configure a User D/extension to dial automatically when the phone is taken
offhook.
S6sl-8 SerFer The P address or URL of System log server. This feature is especially useful for
TSPs.
Grandstream Networks, nc. GXP1400/1405 User Manual Page 23 of 35
Firmware version: 1.0.1.73 Last Updated: 07/2011
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 24 of 35
Last Updated: 07/2011
S6sl-8 LeFel Select the ATA to report the log level. Default is NONE. The level is one of DEBUG,
NFO, WARNNG or ERROR. Syslog messages are sent based on the following
events:
product model/version on boot up (NFO level)
NAT related info (NFO level)
sent or received SP message (DEBUG level)
SP message summary (NFO level)
inbound and outbound calls (NFO level)
registration status change (NFO level)
negotiated codec (NFO level)
Ethernet link up (NFO level)
SLC chip exception (WARNNG and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. n addition to standard Syslog payload, it contains
the following components: $./L0$: 1device &2C address31error code3 error
message.
F or e am p !e: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000].
Ethernet link is up.
Sen0 SIP L-8 When setting the "Yes, phone will send out SP Log to syslog server. Default
setting is "No.
NTP serFer This parameter defines the UR or P address of the NTP (Network Time Protocol)
serve. t is used to display the current date/time.
All-B D9CP O./*-n 4!
/- -Ferr*0e NTP serFer
Default is "Yes. This allows device gets provisioned for DHCP Option 42 from the
server automatically.
SSL Cer/*1*,a/e This defines the SSL certificate needed to access certain websites.
SSL Pr*Fa/e Ie6 This defines the SSL Private key.
SSL Pr*Fa/e Ie6
PassB-r0
This defines the SSL private key password.
D*s/*n,/*Fe %*n8 T-ne Caller D must be configured. Select a Distinctive Ring Tone 1 through 3 for a
particular Caller D. The GXP1400/1405 will ONLY use selected ring tones for
particular Caller Ds. For all other calls, the GXP1400/1405 will use System Ring
Tone. When selected and no Caller D is configured, the selected ring tone will be
used for all incoming calls.
S6s/e) %*n8 T-ne System ring tone. Default is North American standard.
Adjust system ring tone frequencies and cadences based on local
telecom standard.
Call Pr-8ress T-nes Using these settings, users can configure ring or tone frequencies based on
parameters from local telecom. By default, they are set to North American
standard. Frequencies should be configured with known values to avoid
uncomfortable high pitch sounds.
S6n/aL: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing ("On time in 'ms') while OFF is the period of silence.
n order to set a continuous ring, OFF should be zero. Otherwise it will ring ON
ms
and a pause of OFF ms and then repeat the pattern. Up to three cadences are
supported.
D*sa<le Call &a*/*n8 Default is "No. f set to "Yes, the call waiting feature will be disabled.
D*sa<le Call
&a*/*n8 T-ne
Default is "No. f set to "Yes, the call waiting tone will be disabled.
D*sa<le D*re,/ IP Calls Default is "No. f set to "Yes, direct P calls will be disabled.
Use "u*,J IP Call M-0e Dial an P address under the same LAN/VPN segment by entering the last octet in
the P address.
n the Advanced Settings page there is an option "Use Quick P-call mode. Default
setting is "No. When set to "Yes, and #XXX is dialed, where X is 0-9 and XXX
<=255, phone will make direct P call to aaa.bbb.ccc.XXX where aaa.bbb.ccc
comes from the local P address REGARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK). See "u*,J IP Call
M-0e for details.
D*sa<le C-n1eren,e Default is "No. f set to "Yes, conference will be disabled.
D*sa<le DND Bu//-n Default is "No. f set to "Yes, the "DND button on keypad will be disabled.
D*sa<le Trans1er Default is "No. f set to "Yes, transfer will be disabled.
Au/-GA//en0e0 Trans1er Default is "No. f set to "Yes, the phone will use attended transfer by default.
C-n1*8ura/*-n F*a
Ie6.a0 Menu
Configures the access control of configurations via the phone keypad menu. There
are three modes:
Unrestricted
Basic Settings Only
Constraint Mode
D*s.la6 Lan8ua8e Allows user to choose preferred display language in web U and key pad U
Currently, the phone supports these languages: English, Simplified Chinese,
Traditional Chinese, Korean, Japanese, talian, Spanish, French, German,
Portuguese, Russian, Croatian, Hungarian, Polish and Slovenian.
N-/e= The "Automatic setting in language refers to Grandstream's P2Location
client which when connected to nternet would attempt to lookup a database
(driven by Grandstream) with the P address for its geographical location.
Language file postfix allows the language file to have different postfixes so the
phone can request a particular file. t will append an underscore "_" plus the string
in the language file postfix.
The default language file name is "gxp.txt". f the field "Language File postfix "has
"NL" string in it, then the phone will request "gxp_NL.txt" instead of "gxp.txt".
User can only load one secondary language.
Supported downloadable language: Czech, Dutch, Estonian, French, German,
talian, Polish, Portuguese, Slovak, Slovenian and Spanish.
How to set up Download Language:
This is similar to updating firmware in your local network environment.
1. Get the language file gxp.txt ready. Make sure the file is using UTF-8 encoding.
2. Copy gxp.txt to the firmware server directory using your local TFTP or HTTP
server.
3. Access the advanced settings of the Web GU, set "Display Language to
"Download Language and enter the server path in Firmware Server Path. Select
TFTP or HTTP for firmware upgrade.
4. Update and reboot the phone.
Ta<le 1C= SIP A,,-un/ Se//*n8s
A,,-un/ A,/*Fe This field indicates whether the account is active. The default value is "Yes.
A,,-un/ Na)e The name associated with each account - displayed on LCD.
SIP SerFer SP Server's P address or Domain name provided by VoP service provider.
Se,-n0ar6 SIP SerFer This field allows administrator to configure a backup SP Server.
Ou/<-un0 Pr-L6 P address or Domain name of Outbound Proxy, Media Gateway, or Session Border
Controller. Used for firewall or NAT penetration in different network environment. f
the system detects symmetric NAT, STUN will not work. ONLY outbound proxy can
provide solution for symmetric NAT.
SIP User ID User account information provided by VoP service provider (TSP); either an actual
phone number or formatted like one.
Au/Aen/*,a/e ID SP service subscriber's Authenticate D used for authentication. t can be identical
to or different from SP User D.
Au/Aen/*,a/e PassB-r0 SP service subscriber's account password for GXP1400/1405 to register to (SP)
servers of TSP.
Na)e SP service subscriber's name that is used for Caller D display.
DNS M-0e The default is set to A Record. f user wishes to locate the server by DNS SRV,
the user may select SRV or NATPTR/SRV. When "Use Configured P" option is
selected, if SP server is configured as domain name, phone will not send DNS
query, but use "Primary P" or "Secondary P" to send sip message if at least
one of them are not empty.
Pr*)ar6 IP This option applies only if "Use Configured P is selected, the phone will send DNS
query to the Primary P. nsert P address here.
Ba,Ju. IP 1 nsert the first back up P here.
Ba,Ju. IP ! nsert the second back up P here.
SIP %e8*s/ra/*-n This parameter controls sending REGSTER messages to the proxy server.
The default setting is "Yes.
Unre8*s/er -n %e<--/ Default is "No. f set to "Yes, the SP user's registration information will be
cleared on reboot.
%e8*s/er EL.*ra/*-n This parameter allows user to specify the time frequency (in minutes) that
GXP1400/1405 refreshes its registration with the specified registrar. The default
interval is 60 minutes. The maximum interval is 65,535 minutes (about 45
days).
L-,al SIP P-r/ This parameter defines the local SP port used to listen and transmit. The
default value for Account 1 is 5060. t is 5062, 5064, 5066 for Account 2,
Account 3 and Account 4 respectively.
SIP %e8*s/ra/*-n Fa*lure
%e/r6 &a*/ T*)e
Retry registration if the process failed. Default is 20 seconds.
SIP T1 T*)e-u/ RFC 3261 SP T1 timer. Default is 0.5 second.
SIP T! In/erFal RFC 3261 SP T2 timer. Default is 4 seconds.
SIP Trans.-r/ Choose SP Transport between UDP and TCP. Default is UDP.
CAe,J D-)a*n
Cer/*1*,a/e
Enable to check the domain certificate. Default is "No.
%e)-Fe OBP 1r-)
%-u/e
The SP Extension notifies the SP server that it is behind a NAT/firewall.
Val*0a/e In,-)*n8
Messa8es
This configuration selects whether or not the incoming messages should
be validated.
Su..-r/ SIP Ins/an,e ID Selects whether or not SP nstance D is supported.
NAT TraFersal This parameter activates the NAT traversal mechanism. t has options: No,
STUN, Keep-Alive, UPnP, Auto, VPN.
f selecting STUN and a STUN server is also specified, the phone performs
according to the STUN client specification. Using this mode, the embedded STUN
client detects if and what type of NAT/Firewall configuration is used. f the
detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the
phone will use its mapped public P address and port in all of its SP and SDP
messages.
f selecting Keep-Alive with no specified STUN server, the GXP1400/1405 will
periodically (every 20 seconds or so) send a blank UDP packet (with no
payload data) to the SP server to keep the "hole on the NAT open.
SUBSC%IBE 1-r M&I= Default is "No. When set to "Yes a SUBSCRBE for Message Waiting
ndication will be sent periodically.
PUBLIS9 1-r Presen,e Enable Presence feature.
Pr-L6G%e>u*re SP Extension to notify SP server that the unit is behind the NAT/Firewall.
V-*,e Ma*l UserID When configured, user can access messages by pressing "MSG button. This D
is usually the VM portal access number.
Sen0 DTMF This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes: in audio which means DTMF is combined in audio signal
(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SP NFO.
DTMF Pa6l-a0 T6.e Sends DTMF using RFC2833. The default is 101.
Earl6 D*al Default is "No. Use only if proxy supports 484 responses.
D*al Plan Pre1*L Sets the prefix added to each dialed number.
D*al Plan Dial Plan Rules:
1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d
2. Grammar: x - any digit from 0-9;
a) xx+ - at least 2 digit numbers
b) xx. - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
e) [147] - any digit of 1, 4, or 7
f) <2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617
Example 2: {^1900x+ | <=1617>xxxxxxx}
Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit
numbers
Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}
Allows any number with leading digit 1 followed by a 3 digit number, followed by any
number between 2 and 9, followed by any 7 digit number OR Allows any length of
numbers with leading digit 2, replacing the 2 with 011 when dialed.
3. Default: Outgoing {x+}
Allow any length of numbers.
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):
^1900x. - prevents dialing any number started with 1900
<=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing
7 numbers and 1617 area code will be added automatically
1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits
length
011[2-9]x. - allows international calls starting with 011
[3469]11 - allow dialing special and emergency numbers 311, 411, 611 and 911
N-/e= n some cases where the user wishes to dial strings such as *123 to activate
voice mail or other applications provided by their service provider, the * should be
predefined inside the dial plan feature. An example dial plan will be: { *x+ } which
allows the user to dial * followed by any length of numbers.
Dela6e0 Call F-rBar0
&a*/ T*)e
Time waited before the call is forward to a number or VM.
Default is 20 seconds.
Ena<le Call Fea/ures Default is "Yes. f set to "No, Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided TSP support those features. n addition, "ForwardAll
softkey will be hidden if call feature code is disabled for Account 1.
Call L-8 User can choose to disable Call Log and what kind of calls to log.
Grandstream Networks, nc. GXP1400/1405 User Manual Page 29 of 35
Firmware version: 1.0.1.73 Last Updated: 07/2011
Sess*-n EL.*ra/*-n The SP Session Timer extension enables SP sessions to be periodically
"refreshed via a SP request (UPDATE, or re-NVTE. Once the session interval
expires, if there is no refresh via a UPDATE or re-NVTE message, the session
is terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
M*nGSE Defines the minimum session expiration (in seconds). Default is 30 se,-n0s.
Caller %e>ues/ T*)er f set to "Yes, the phone will use session timer when it makes outbound calls
if remote party supports session timer.
Callee %e>ues/ T*)er f selecting "Yes, the phone will use session timer when it receives inbound
calls with session timer request.
F-r,e T*)er f set to "Yes, the phone will use session timer even if the remote party does not
support this feature. f set to "No, the session timer is enabled only when the
remote party supports this feature. To turn off Session Timer, select "No for
Caller Request Timer, Callee Request Timer, and Force Timer.
UAC S.e,*16 %e1resAer As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
UAS S.e,*16 %e1resAer As a Callee, select UAC to use caller or proxy server as the refresher, or UAS
to use the phone as the refresher.
F-r,e INVITE Session Timer can be refreshed using NVTE method or UPDATE method. Select
"Yes to use NVTE method to refresh the session timer.
Ena<le 100rel PRACK (Provisional Acknowledgment) method enables reliability to SP
provisional responses (1xx series). This is required to support PSTN inter-
networking.
A,,-un/ %*n8 T-ne There are 4 uniquely defined ring tones:
One (1) System Ring Tone: when selected, all calls will ring with system
ring tone.
Three (3) Customer Ring Tones: when selected, incoming calls from
designated account will play selected ring tone.
%*n8 T*)e-u/ Defines how long ring will ring when receiving a call. Default is 60 seconds.
Sen0 An-n6)-us f this parameter is set to "Yes, the "From header in outgoing NVTE message
will be set to anonymous, essentially blocking the Caller D from displaying.
An-n6)-us Call
%e?e,/*-n
Default is "No. f set to "Yes, anonymous call will be rejected.
Au/- AnsBer Default is "No. f set to "Yes, GXP1400/1405 will automatically switch on
speaker to answer the incoming call. Set to ntercom/Paging mode, it will answer
the call based on the SP info header from the server.
All-B Au/- AnsBer <6
CallGIn1-
f the Call-nfo header contains answer-after=0, the call be answered automatically
(so called paging mode).
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Firmware version: 1.0.1.73 Last Updated: 07/2011
%e1erGT- Use Tar8e/
C-n/a,/
Default is "No. f set to "Yes, then for Attended Transfer, the "Refer-To header
uses the transferred target's Contact header information.
Trans1er -n C-n1eren,e
9an8u.
Defines whether or not the call is transferred to the other party if the initiator of the
conference hangs up.
Default setting is set to "No.
Pre1erre0 V-,-0er GXP1400/1405 supports up to 7 different Vocoder types including G.711(a/) (also
known as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).
Configure Vocoders in a preference list that is included with the same preference
order in SDP message. Enter the first Vocoder in this list by choosing the
appropriate option in "Choice 1. Similarly, enter the last Vocoder in this list by
choosing the appropriate option in "Choice 8.
S%TP M-0e Enable SRTP mode based on selection. Default is "No.
S6))e/r*, %TP Selects whether or not symmetric RTP is supported.
S*len,e Su..ress*-n This controls the silence suppression/VAD feature of the audio codec G.723 and
G.729. f set to "Yes, when silence is detected, a small quantity of VAD packets
(instead of audio packets) will be sent during the period of no talking. f set to "No,
this feature is disabled.
V-*,e Fra)es .er TX This field contains the number of voice frames to be transmitted in a single Ethernet
packet (be advised the S limit is based on the maximum size of Ethernet packet is
1500 byte (or 120kbps)).
When setting this value, be aware of the requested packet time (ptime, used in SDP
message) is a result of configuring this parameter. This parameter is associated
with the f irs t codec in the above codec Preference List or the actual used payload
type negotiated between the 2 conversation parties at run time. 4.g., if the first
codec is configured as G.723 and the "Voice Frames per TX is set to 2, then the
"ptime value in the SDP message of an NVTE request will be 60 m s because each
G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the
first codec is G.729 or G.711 or G.726, then the "ptime value in the SDP message
of an NVTE request will be 20 m s .
f the configured voice frames per TX exceeds the maximum allowed value, the P
phone will use and save the maximum allowed value for the corresponding first
codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20
(x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms)
and 64 (x2.5ms) frames respectively.
Please be careful when editing these parameters. Adjusting these parameters will
also change the dynamic jitter buffer. The GXP1400/1405 has a patent dynamic
jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.
We recommend using the default settings provided. We do not recommend
adjusting these parameters if you are an average user. ncorrect settings will affect
the voice quality.
N- Ie6 En/r6 T*)e-u/ Default is 4 seconds.
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Use O as D*al Ie6 This parameter allows users to configure the "# key as the "Send (or "Dial) key.
f set to "Yes, the "# key will immediately send the call. n this case, this key is
essentially equivalent to the "(Re)Dial key. f set to "No, the "# key is included as
part of the dial string.
GD!5 %a/e Encoding rate for G723 codec. By default, 6.3kbps rate is set.
GD!CG5! Pa,J*n8 M-0e Select "TU or "ETF for G726-32 packing mode.
*LBC Fra)e S*+e iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might
be required.
*LBC Pa6l-a0 T6.e Payload type for iLBC. Default value is 97. The valid range is between 96 and 127.
S.e,*al Fea/ure Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.
SAVING T9E CONFIGU%ATION C9ANGES
After the user makes a change to the configuration, press the "Update button in the Configuration Menu.
The web browser will then display a message window to confirm saved changes.
We recommend rebooting or powering cycle the P phone after saving
changes.
%EBOOTING T9E P9ONE %EMOTEL'
Press the "Reboot button at the bottom of the configuration menu to reboot the phone remotely. The web
browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in
again.
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Actualizacin y personalizacin de software
Software (firmware) o las actualizaciones se realizan a travs de TFTP o HTTP. Los valores
de configuracin correspondientes se encuentran en la pgina de configuracin Ajuste
avanzado.
ACTUALZACN FRMWARE TRAVS TFTP / HTTP
Para actualizar a travs de TFTP o HTTP, seleccione TFTP o mtodo de actualizacin HTTP.
"Servidor de actualizacin" se debe establecer a una URL vlida de un servidor HTTP.
Nombre del servidor puede estar en formato de direccin P o FQDN. Estos son ejemplos de
algunas de las URL vlidos.
firmware.mycompany.com:6688/Grandstream/1.2.3.5
72.172.83.110
Hay dos formas de configurar el servidor de actualizacin para actualizar el firmware: a
travs de la tecla de men Pad y Web
nterfaz de configuracin.
Tecla Men Pad
Para configurar el servidor de actualizacin a travs de las opciones de tecla de men Pad,
seleccione "Configuracin" en el men principal, seleccione "Actualizacin". Bajo este sub
men, el usuario puede editar servidor de actualizacin, ya sea en un formato de direccin
P o el formato FQDN. Seleccione la opcin "Guardar y utilizar TFTP" o "Guardar y utilizar
HTTP" para seleccionar el mtodo de actualizacin. Seleccione "Reiniciar" en el men
principal para reiniciar el telfono.
nterfaz de configuracin Web
Para configurar el servidor de actualizacin a travs de la interfaz de configuracin Web,
abra el navegador web. ntroduzca la direccin P GXP1400 / 1405. ntroduzca la contrasea
de administrador para acceder a la interfaz web de configuracin. En la pgina Ajustes
avanzados, introduzca la direccin P del servidor de actualizacin o FQDN en el campo
"Ruta del servidor de firmware". Seleccione TFTP o HTTP mtodo de actualizacin.
Actualizar el cambio haciendo clic en el botn "Actualizar". "Reboot" o apagar y encender el
telfono para actualizar el nuevo firmware.
Durante esta etapa, la pantalla LCD mostrar el proceso de descarga de archivos de
firmware. Por favor, no interrumpir o apagar la unidad. Si una actualizacin de firmware falla
por cualquier motivo (por ejemplo, servidor TFTP / HTTP no responde, no hay archivos de
imgenes de cdigos est disponible para la actualizacin, o no supera la prueba de suma
de comprobacin, etc), el telfono se detendr el proceso de mejora y re-arranque usando el
firmware / software existente.
Las actualizaciones de firmware toman alrededor de 60 segundos en una LAN controlada o
5-10 minutos a travs de nternet. Le recomendamos completar las actualizaciones de
firmware en un entorno LAN controlada siempre que sea posible.
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No TFTP Local / HTTP Server
Para los usuarios que no tienen un servidor TFTP / HTTP locales, proporcionamos un
servidor HTTP en la nternet pblica para que los usuarios descarguen la ltima
actualizacin del firmware de forma automtica. Por favor, consulte la seccin Soporte /
Descarga de nuestro sitio web para obtener esta direccin P del servidor HTTP:
http://www.grandstream.com/support/firmware.
Alternativamente, descargue e instale un TFTP libre o HTTP a la LAN para realizar
actualizaciones de firmware. A
servidor gratuito de Windows versin TFTP est disponible:
http://support.solarwinds.net/updates/New-customerFree.cfm.

NSTRUCCONES PARA ACTUALZAR TFTP LOCAL:
1. Descomprimir el archivo y poner todos ellos en el directorio raz del servidor TFTP.
2. La PC que ejecuta el servidor TFTP y el GXP1400 / 1405 debe estar en la misma LAN
segmento.
3. Vaya a Archivo -> Configurar -> Seguridad para cambiar la configuracin predeterminada
del servidor TFTP desde
"Slo recepcin" a "Transmitir Solamente" para la actualizacin del firmware.
4. nicie el servidor TFTP, en la pgina web de configuracin del telfono
5. configurar la ruta del servidor de firmware con la direccin P de la PC
6. Actualizar el cambio y reiniciar la unidad
El usuario tambin puede optar por descargar el servidor HTTP libre de
http://httpd.apache.org/ o utilizar Microsoft S
servidor web.
Nota:
Cuando GXP1400 / 1405 telfono se inicia, enviar TFTP o solicitud HTTP para descargar
el archivo de configuracin "cfg000b82xxxxxx", donde "000b82xxxxxx" es la direccin MAC
de la / 1405 telfono GXP1400. Este archivo es para el aprovisionamiento de propsito. Para
las actualizaciones normales TFTP o HTTP de firmware, los siguientes mensajes de error en
un registro del servidor TFTP o HTTP se pueden ignorar: "Error de TFTP [P DRECCN]
solicitando cfg000b82023dd4: El archivo no existe. Configuracin de descarga de archivos "
DESCARGAR ARCHVO DE CONFGURACN
El GXP1400 / 1405 se puede configurar a travs de la nterfaz Web, as como a travs del
archivo de configuracin (binario o XML) a travs de TFTP o HTTP / HTTPS. La "Ruta del
servidor Config" es la ruta del servidor TFTP o HTTP para el archivo de configuracin. Tiene
que ser ajustado a una URL vlida, ya sea en formato de direccin P o FQDN. La "Ruta
Config Server" puede ser el mismo o diferente de la "Ruta del servidor de firmware".
Un parmetro de configuracin se asocia a cada campo en particular en la pgina de
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configuracin web. Un parmetro consiste en una letra mayscula P y nmeros numricas 2
a 4 dgitos. es decir, P2 se asocia con "Contrasea de administrador" en la pgina Ajustes
avanzados. Para obtener una lista de parmetros detallada, por favor refirase a la plantilla
de configuracin correspondiente del firmware.
Una vez que los GXP1400 / 1405 arranca (o reiniciado), se solicitar el archivo de
configuracin llamado "cfgxxxxxxxxxxxx" seguido de una solicitud de archivo XML de
configuracin llamado "cfgxxxxxxxxxxxx.xml", donde "xxxxxxxxxxxx" es la direccin MAC del
dispositivo , es decir, "cfg000b820102ab". El nombre del archivo de configuracin debe estar
en minsculas.
Para ms detalles sobre el aprovisionamiento XML, consulte
http://www.grandstream.com/support.
Gestin de firmware y configuracin de descarga de archivos
Cuando "Actualizacin automtica" est ajustado a "S", un proveedor de servicios puede
utilizar P193 (Auto Check intervalo, en minutos, por defecto y el mnimo es de 60 minutos)
para que los dispositivos de comprobar peridicamente si hay actualizaciones en intervalos
de tiempo pre-programado. Al definir diferentes intervalos en P193 para diferentes
dispositivos, un proveedor Server puede gestionar y reducir el firmware o la carga del
servidor de aprovisionamiento en un momento dado.

Restaurar Valores predeterminados de fbrica
ADVERTENCA: Al restaurar el valor predeterminado de fbrica borrar toda la informacin
de configuracin del telfono. Por favor respalde o imprimir todos los ajustes antes de
restablecer la configuracin de fbrica. No somos responsables de la restauracin de los
parmetros perdidos y no podemos conectar el dispositivo a su proveedor de servicios de
VoP.
NSTRUCCONES PARA LA RESTAURACN:
Paso 1: Pulse el botn "OK" para que aparezca el men de configuracin del teclado,
seleccione "Configuracin" y pulse "OK" para acceder al submen, seleccionar "Factory
Reset" (Por favor, consulte la Tabla 5-1 del diagrama de flujo teclado)
Paso 2: ntroduzca la direccin MAC impresa en la parte inferior de la etiqueta. Utilice el
siguiente mapeo:
0-9: 0-9
A: 22 (presione la tecla "2" dos veces, "A" se mostrar en la pantalla LCD) B: 222
C: 2222
D: 33 (pulse la tecla "3" dos veces, "D" se mostrar en la pantalla LCD) E: 333
F: 3333
Ejemplo: si la direccin MAC es 000b8200e395, debe ser clave en como
"0002228200333395".
Grandstream Networks, nc. GXP1400/1405 User Manual
Firmware version: 1.0.1.73
Page 36 of 35
Last Updated: 07/2011
Nota:
Si existen dgitos como "22" en el MAC, tienes que escribir "2" y pulse "->" tecla de flecha
derecha para mover el cursor o esperar durante 4 segundos a que sigan clave en otro "2".
Paso 3: Pulse el botn "OK" para mover el cursor a "OK". Pulse el botn "Aceptar" de nuevo
para confirmar. Si el
La direccin MAC es correcto, el telfono se reiniciar. De lo contrario, saldr a la anterior
interfaz de men del teclado.

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