Seltsames Soundproblem mit 7940

[klinke]

Hallo zusammen,

ich spiele hier mit einem Asterisk und nem 7940er rum. SIP Calls funktionieren einwandfrei in guter Qualität. Interessanterweise wird die Qualität aber sau schlecht, sobald ich über ISDN raus oder reinrufe. Nun würde man denken, dass das Problem irgendwo bei * liegt. Was es aber wohl nicht tut, da wenn ich ein Softphone benutze die Telefonate raus und rein in einer guten Qualität gehen. Ich kann es mir irgendwie nicht erklären und habe auch schon einiges ausprobiert. Es muss was mit dem 7940 zu tun haben.
Aktuell ist es mit einer SIP Firmware Version 6.2 geflascht (würde es hier Sinn machen zu updaten?).

Und ich nutze folgende Einstellungen am 7940:
SIPDefault.cnf:

P Default Generic Configuration File
########################################
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
#image_version: P0S3-06-2-00

# Proxy Server
proxy1_address: ""              ; 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: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

# 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: g711alaw

# 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: "208.69.32.139"                    ; SNTP Server IP Address
sntp_mode: unicast                   ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET                         ; 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: Sunday   ; Day of week in which DST starts
dst_start_week_of_month: 1      ; Week of month in which DST starts
dst_start_time: 02                     ; 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: 2                        ; 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: ""                ; Dotted IP of Backup Proxy
proxy_backup_port: 5060         ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: ""             ; 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: 5060         ; UDP port used for SIP messages (default - 5060)
start_media_port: 5004          ; Start RTP range for media (default - 16384)
end_media_port: 5007            ; End RTP range for media (default - 32766)
nat_received_processing: 1      ; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: ""              ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060       ; 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: 1       ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 1                 ; 0-Disabled (default), 1-Enabled, 2-Privileged

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

# XML URLs
services_url: "http://www.fo-pa.de/cgi-bin/rss2cisco.pl"                ; URL for external Phone Services
directory_url: "//niels/tftp-root/directory.xml"                ; URL for external Directory location
logo_url: "http://192.168.99.9/asterisk/asterisk-tux.bmp"                       ; URL for branding logo to be used on phone display

# 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: ""               ; 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: 0           ; Default 0 (Call Hold Ringback feature is off)

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

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

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

# Telefonnummern automatisch vervollstaendigen (macht bei mir Probleme, also aus)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
autocomplete: 0                 ; 0-Disabled, 1-Enabled

und

SIPMac.conf:

# SIP Configuration Generic File

phone_label: "CISCO 7940"

######################################### Line 1 -

proxy1_address: "192.168.99.9"
proxy1_port: 5060
line1_name: "31"
line1_authname: "31"
line1_password: "31"
line1_displayname: "Line 1"
line1_shortname: "Line 1"


######################################### Line 2

proxy2_address: "192.168.99.9"
proxy2_port: 5060
line2_name: "32"
line2_authname: "32"
line2_password: "32"
line2_displayname: "Line 2"
line2_shortname: "Line 2"

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

# All user_parameters have been removed

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

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "Friedrich <31>"

# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: "Friedrich <32>"

####### New Parameters added in Release 3.0 ######

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt:   "SIP Phone"      ; Limited to 15 characters (Default - SIP Phone)

# Phone Password (Password to be used for console or telnet login)
phone_password: "Cisco" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none

 

[janthedoc81]

Hallo,

probier doch mal die neuste Firmware, habe ich auch auf meinem 7940 laufen und keine Probleme. Vielleicht bringts was.

Bei preferred_codec: g711alaw kannst Du mal none probieren, dann nimmt er jeden Codec...

Ansonsten kann man per Ferndiagnose wenig sagen, hast Du noch ein anderes Telefon zum testen?

Gruss

jan