Cisco phone (sip прошивка) и mtt - youmagic, не работает входящая связь

ставите хаб перед маршрутизатором, снифаете сип-трафик, смотрите, кто косячит

в конфигурации циско телефона(которую он грузит по tftp) поставьте нат.с клавиатуры телефона нат в 1 вам никак не получиться поставить, только по tftp.

вот полный рабочий конфиг для 7940

SIPDefault.cnf

# Image Version
image_version: "P0S3-8-12-00"

# Proxy Server
# Note: I put the proxy server information in the individual conf files
# for each machine, since each box has different configs.  You could, of course,
# put all of them here in the Default file...
proxy1_address: "78.46.118.22"
preferred_codec:g711alaw
proxy1_port:"5060"

# Proxy Server Port (default - 5061)
# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: ""

# NAT/Firewall Traversal
nat_enable: "1"
nat_address: ""
# change to different port for every phone
voip_control_port: "5061"
start_media_port: "16384"
end_media_port:  "32766"
nat_received_processing: "1"

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "120"

# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "none"

# 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: "1"

# 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: "*98"

#*********  Release 2 new config parameters **********

# TFTP Phone Specific Configuration File Directory

# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "17.254.0.49"
time_zone: "EEST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: "1"

# 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: "1"

####### New Parameters added in Release 4.0 #######

# XML URLs
#services_url: "http://192.168.1.145/menu.pl" ; URL for external Phone Services
#services_url: "http://192.168.1.254/cgi-bin/cisco7960/menu.xml"  ; test url
#directory_url: "http://192.168.7.101/voip_xml_utils/index.cfm?fuseaction=dspEmployeeMenu"               
# URL for external Directory location

logo_url: "http://78.46.118.22/asterisk-tux.bmp"
# put your own logo in the logo_url location; I include the 10-20.com one for reference in building your own

# HTTP Proxy Support
http_proxy_addr: ""             ; Address of HTTP Proxy server
http_proxy_port: 80             ; Port of HTTP Proxy Server (80-default)

# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""              ; restricted to dotted IP
dyn_dns_addr_2: ""              ; restricted to dotted IP
dyn_tftp_addr: "192.168.3.1"   ; restricted to dotted IP

# The dynamic tftp server should be set to whatever your TFTP server is.  This way, it
# keeps the tftp server setting even though you might be using DHCP (default behavior
# is to use the DHCP server as a tftp server, which is rarely correct.)

# Remote Party ID
remote_party_id: 1              ; 0-Disabled (default), 1-Enabled

SIP000XXXXXXXXX.cnf(заменить на адрес телефона)

# Line 1 Settings
 line1_name: "12"                     ; Line 1 Extension\User ID

 line1_displayname: "12"           ; Line 1 Display Name
 line1_shortname: "12"      ; Comment next to the button
 line1_authname: "12"         ; Line 1 Registration Authentication
 line1_password: "111111"         ; Line 1 Registration Password