cisco 7940 and kleines 'x' (unregistered?) A@H 2.8

[frickelmicha]

setup:

1 x A@H 2.8 hinter NAT router (IP=192.168.0.10)
1 x Cisco Phone 7940G mit SIP V 8.2 firmware (192.168.0.20), extension 102
1 x XTEN softphone (192.168.0.100)
1 x Digium TDM400P mit 1 x FXO und 1 x FXS
1 x analoges Telefon, extension 101
1 x U.K. pstn line
1 x sipgate (UK)

Hi,

Habe das Problem, dass mein Cisco 7940G ( SIP Version 8.2 ) nicht ganz richtig mit A@H 2.8 funktioniert.

Symptome sind:

- kleines 'x' ganz rechts im display (extension 102)

- trotzdem zeigt mir das web interface 'Asterisk Info' unter SIP Registry an,
dass das Cisco registriert ist ( mit Status = OK (175 ms). Warum eigentlich
solch eine lange zeit? SIPGATE ist um die 50ms und ein XTEN softphone im
selben lokalen next ist 2 ms.

- kann mit dem cisco (ext 102) nicht mein analoges phone (ext 101) anrufen.
Bekomme von a@h: 'Person on ext 101 is unavailable'

- umgekehrt vom analogen phone zum Cisco phone gibt's dieselbe Ansage
(101 und 102 natuerlich vertauscht)

- Anrufe vom pstn werden von A@H angenommen und zur analogen extension
101 weitergeleitet

- kann leider nicht rauswaehlen (weder pstn oder sipgate, sipgate's website sagt
mir, dass ich online bin )

- softphone zeigt mir im web interface ebenso an, dass es registriert ist,
bekomme aber noch nicht einmal A@H Ansagen, wie z.B. clock: *60.
Im X-Lite display steht: 'Call failed: Error 403 forbidden'

Hier noch meine configuration files:

SIPDefault.cnf :

# SIP default configuration file for Cisco 79xx 
# for use in the U.K. (relevant only for time settings)

# Image Version
# put in the phone's current firmware version in the next line:
image_version: P0S3-08-2-00

# Proxy Server
proxy1_address: "192.168.0.10"  ; Can be dotted IP or FQDN
proxy2_address: ""		; Can be dotted IP or FQDN
proxy3_address: ""		; Can be dotted IP or FQDN
proxy4_address: ""		; Can be dotted IP or FQDN
proxy5_address: ""		; Can be dotted IP or FQDN
proxy6_address: ""		; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: 5060 
proxy2_port: ""
proxy3_port: ""
proxy4_port: ""
proxy5_port: ""
proxy6_port: ""

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

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 500

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: none

# TOS bits in media stream [0-5] (Default - 5)
#tos_media: "5"

# 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 10
sip_invite_retx: 6 		; Default 6
timer_invite_expires: 180 	; Default 180 sec

####### New Parameters added in Release 2.0 #######

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: ""		; Example:  ./sip_phone/
 
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.168.0.10"	; SNTP Server IP Address
sntp_mode: unicast		; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: GMT			; Time Zone Phone is in
dst_offset: 1			; Offset from Phone's time when DST is in effect 
dst_start_month: April		; Month in which DST starts
dst_start_day: ""		; Day of month in which DST starts
dst_start_day_of_week: Sun	; Day of week in which DST starts
dst_start_week_of_month: 8	; Week of month in which DST starts
dst_start_time: 01		; Time of day in which DST starts
dst_stop_month: Oct		; Month in which DST stops
dst_stop_day: ""		; Day of month in which DST stops
dst_stop_day_of_week: Sunday	; Day of week in which DST stops
dst_stop_week_of_month: 8	; Week of month in which DST stops 8=last week of month
dst_stop_time: 02		; Time of day in which DST stops
dst_auto_adjust: 1		; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1		; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format: D-M-YY		; Dateformat Day, month, year

# 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-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0		; Default 0 (Disable blocking of anonymous calls)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101		; Default 101

# Sync value of the phone used for remote reset 
sync: 1				; Default 1

####### New Parameters added in Release 2.1 #######

# Backup Proxy Support
proxy_backup: "192.168.0.10"	; Dotted IP of Backup Proxy
proxy_backup_port: 5060		; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "192.168.0.10" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060	; Emergency Proxy port (default is 5060)

# Configurable VAD option
enable_vad: 0			; VAD setting 0-disable (Default), 1-enable

####### New Parameters added in Release 2.2 ######

# NAT/Firewall Traversal
nat_enable: "0"			; 0-Disabled (default), 1-Enabled
nat_address: ""			; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: ""		; UDP port used for SIP messages (default - 5060)
start_media_port: ""		; Start RTP range for media (default - 16384)
end_media_port: ""	   	; End RTP range for media (default - 32766)
nat_received_processing: "0"	; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: ""	 	; restricted to dotted IP or DNS A record only
outbound_proxy_port: ""		; default is 5060

####### New Parameter added in Release 3.0 #######

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1		; 0-Disabled, 1-Enabled (default)

####### New Parameters added in Release 3.1 #######

# 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 the phone) 
telnet_level: 2			; 0-Disabled (default), 1-Enabled, 2-Privileged

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

# URL for external Phone Services
services_url: "http://192.168.0.10/cisco/services/index_cisco.php"

# URL for external Directory location
directory_url: "http://192.168.0.10/cisco/services/PhoneDirectory.php"

# URL for branding logo
logo_url: "http://192.168.0.10/cisco/bmp/aah.bmp"

# HTTP Proxy Support
#http_proxy_addr: ""		; Address of HTTP Proxy server
#http_proxy_port: ""		; 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: ""              ; restricted to dotted IP

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

####### New Parameters added in Release 4.4 #######

# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 1		; Default 0 (Call Hold Ringback feature is off)

####### New Parameters added in Release 6.0 #######

# Dialtone Stutter for MWI (Message Waiting Indicator)
stutter_msg_waiting: 1         ; 0-Disabled (default), 1-Enabled

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 1                  ; 0-Disabled (default), 1-Enabled

# autocomplete telephone numbers when dialling
autocomplete: 0                ;

und SIP00xxxxxx.cnf

# SIP Configuration File (start)

# Line 1 Settings
line1_name:        "102"        ; Line 1 Extension\User ID
line1_shortname:   "102"        ; Line 1 Display Name
line1_displayname: "102"        ; Line 1 CID Display
line1_authname:    "102"        ; Line 1 Registration Authentication
line1_password:    "cisco284w"  ; Line 1 Registration Password

# Line 2 Settings
line2_name:        ""           ; Line 2 Extension\User ID
line2_shortname:   ""           ; Line 2 Display Name
line2_displayname: ""           ; Line 2 CID Display
line2_authname:    ""           ; Line 2 Registration Authentication
line2_password:    ""           ; Line 2 Registration Password

# Line 3 Settings
line3_name:        ""           ; Line 3 Extension\User ID
line3_shortname:   ""           ; Line 3 Display Name
line3_displayname: ""           ; Line 3 CID Display
line3_authname:    ""           ; Line 3 Registration Authentication
line3_password:    ""           ; Line 3 Registration Password

# Line 4 Settings
line4_name:        ""           ; Line 4 Extension\User ID
line4_shortname:   ""           ; Line 4 Display Name
line4_displayname: ""           ; Line 4 CID Display
line4_authname:    ""           ; Line 4 Registration Authentication
line4_password:    ""           ; Line 4 Registration Password

# Line 5 Settings
line5_name:        ""           ; Line 5 Extension\User ID
line5_shortname:   ""           ; Line 5 Display Name
line5_displayname: ""           ; Line 5 CID Display
line5_authname:    ""           ; Line 5 Registration Authentication
line5_password:    ""           ; Line 5 Registration Password

# Line 6 Settings
line6_name:        ""           ; Line 6 Extension\User ID
line6_shortname:   ""           ; Line 6 Display Name
line6_displayname: ""           ; Line 6 CID Display
line6_authname:    ""           ; Line 6 Registration Authentication
line6_password:    ""           ; Line 6 Registration Password

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Asterisk  "       ; Has no effect on SIP messaging

user_info:         phone        ; for message waiting light

messages_uri:      "*97"        ; configure "messages" button for voice mail

# Phone prompt/password for telnet/console session
phone_prompt:   "SIP"           ; Telnet/Console Prompt
phone_password: "cisco"         ; Telnet/Console Password

# SIP Configuration Generic File (stop)

Ich probiere nun schon seit Tagen an diesem System und habe unzaehlige Forenbeitrage und webseiten gelesen, aber irgendwie komme ich absolut nicht weiter.

Wuerde mich sehr ueber Tips von euch freuen.

Danke schon mal im vorraus.

 

[alex-911]

hi

das 7940/7960 ist nicht ganz ohne bezüglich NAT traversal.

setz mal in deiner sip.conf für dein 7940:

nat=no

ich vermute mal es steht auf "yes"?

gruss
/alex

 

[frickelmicha]

Hi alex,

Danke fuer deine Antwort - dachte schon das Forum hier ist nicht sehr busy... ;-) oder vielleicht ist es auch nur 'ne dumme Frage.

Die a@h box und das Cisco phone sind beide hinter der NAT firewall, daher dachte ich eigentlich nicht, dass ich das brauche...

Werd's morgen mal versuchen.

tschuess

mfg micha