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Yo tengo una centralita en este caso es Elastix con sus extensiones creadas y los tengo integrado con telefonos 7940 de Cisco, y algunos telefonos android con Zoiper, todas mis extensiones son internas y se registran bien, los softphones funcionan a todo dar, pero el problema con mis telefonos cisco es este: a los 30 segundos se vuelve unreachable, la primera llamada antes de los 30 segundos la saca, pero despues de los 30 se vuelve unreachable y ya no vuelve a sacar llamadas, a diferencia de los softphones, siguen funcionando, necesito que alguien me eche la mano, todas son extensiones internas. ya Cambie el Qualify de Yes a No y no funciono, ¿Que es lo que puede estar ocasionando este problema?
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umm extraño , nunca he usado Elastix pero me huele a que podria ser la plantilla que estas usando para skyni o lo tienes parcheado para que use sip generico ese tel? -
No creo, no estoy usando plantillas para Skyni, y no he parcha para uso de generico de SIP, de hecho aqui te pongo las configuraciones de mis telefonos, quiza hay algo mal ahi, las extensiones en el elastix los hice por la via web del conmutador.
Esta es mi version del firmware del Sip P003-8-12-00
este es mi SIPDEFAULT.cnf
#Image Version
image_version: "P0S3-8-12-00"
#Proxy server
proxy1_address: "192.168.1.200"
#Proxy server
proxy2_address: "192.168.1.200"
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
# Emergency Proxy info
proxy_emergency: "192.168.1.200"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "192.168.1.200"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: "192.168.1.200"
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "no"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"
# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "g711ulaw"
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"
# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "1" ; 0-Disabled (default), 1-Enabled, 2-Privileged
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "0"
# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: "avt"
# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up,5-6dB up)
dtmf_db_level: "3"
# SIP Timers
timer_t1: "500" ; Default 500 msec
timer_t2: "4000" ; Default 4 sec
sip_retx: "10" ; Default 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "2500"
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)
# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)
# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100
# XML file that specifies the dialplan desired
dial_template: "dialplan"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "0"
#URL for external Phone services
services_url: "http://192.168.1.200/cisco/services.xml"
# URL for external Directory location
logo_url: "http://192.168.1.200/cisco/directory.xml"
# URL for branding logo
directory_url: "http://192.168.1.200/cisco/asterisk-tux.bmp"
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
Y este es mi mi SIP<MacAddress>.cnf
phone_label: "Gerencia"
line1_name:"4511"
line1_shortname:"4511-1"
line1_displayname:"Gerencia"
line1_password: "45104510MA"
line2_name:"4511"
line2_shortname:"4511-2"
line2_displayname:"Gerencia"
line2_password: "45104510MA"
line3_name:"4511-3"
line3_shortname:"4511-3"
line3_displayname:"Gerencia"
line3_password: "45104510MA"
line4_name:"4511-4"
line4_shortname:"4511-4"
line4_displayname:"Gerencia"
line4_password: "45104510MA"
line5_name: "UNPROVISIONED"
line5_shortname: "UNPROVISIONED"
line5_displayname: "UNPROVISIONED"
line5_password: "UNPROVISIONED"
line6_name: "UNPROVISIONED"
line6_shortname: "UNPROVISIONED"
line6_displayname: "UNPROVISIONED"
line6_password: "UNPROVISIONED"
line1_authname: "4511"
line2_authname: "4511"
line3_authname: "4511"
line4_authname: "4511"
line5_authname: "UNPROVISIONED"
line6_authname: "UNPROVISIONED"
time_zone: "CST"
time_format_24hr: "1"Comentario
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