Trixbox2.6 -SIP raus und eingehende Anrufe funktionieren nicht

[horgee]

Moin,

habe versucht von Asterisk 1.2 auf Trixbox umzustellen.

Leider funktioniert bei mir scheinbar nicht viel. Wenn ich auf dem Cisco eine Nummer eingebe (z.B. *97 oder *43) bekomme ich ein dem Besetztton ähnliches Signal. In der Anzeige steht zu dem
Reorder.
Bei Anrufen von aussen mit dem Mobiltelefon oder per Festnetz bekomme ich den Klingelton (beim anrufenden Telefon), komme aber nicht auf die eingerichtete Voicemail.

Hier meine Konfiguration:
Trixbox 2.6 vom 13.06.08
fritz card pci mit capi
Cisco 7960G , FW P0S3-07-5-00

IP's
Trixbox 192.168.0.107
Cisco Phone 192.168.0.105


Trixbox wollte ich ausprobieren da ich unter Asterisk das mit MOH nicht hinbekommen habe. Asterisk produzierte auch einige Fehlermeldungen, die ich nicht in den Griff bekam.
Da ich eine Flatrate habe nutze ich keinen SIP Provider, sondern ISDN. Ehrlich gesagt, der einzige Grund warum ich Asterisk genutzt habe und Trixbox nutzen will ist das ich das Cisco Telefon einfach toll finde. Habe an das Telefon noch ein Plantronics Headset angeschlossen. Damit macht vieltelefonieren Spass

Was ich gemacht habe:
Nach der Grundinstallation von Trixbox, habe ich nach einer sehr guten Anleitung hier aus dem Forum (http://www.ip-phone-forum.de/showthread.php?t=166154) capi installiert. Capiinfo gibt die korrekte Info aus. Scheint also zu funktionieren.
In der modules.conf habe ich den entsprechenden Eintrag gemacht:

; Joerg, 080702
load => res_features.so
load => chan_capi.so

[global]
chan_capi.so=yes
; Joerg, 080702

Danach habe ich die Inbound Routes, Extensions und Custom Trunk (Custom Dial String: CAPI/ISDN1/$OUTNUM$, Dial Rules: ._XXXX.) angelegt.

Die Extension für das Cisco sieht wie folgt aus:

[10]
type=peer
secret=10
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=1
nat=
mailbox=10@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/10
context=from-internal
canreinvite=yes
callgroup=1
callerid=device <10>
allow=all
accountcode=
call-limit=50

Die /etc/capi.conf

; Joerg, 080702
[ISDN1]     			    
; incomingmsn=* 
incomingmsn=MEINEMSN      ; MEINEMSN ist natürlich eine Nummer
msn=MEINEMSN                 ; "
controller=2
defaultcid=MEINEMSN         ; "
isdnmode=msn
immediate=yes
group=1
devices=2
softdtmf=on
relaxdtmf=on
faxdetect=off
context=default
echocancel=yes
canpark=yes

Die SIP000785056314.cnf des Cisco

# Cisco SIP Configuration

phone_label: "Home Office"
line1_name: "10"
line1_shortname: "10"
line1_displayname: "10"
line1_password: "10"
line2_name: "20"
line2_shortname: "20"
line2_displayname: "20"
line2_password: "20"

Die SIPDefault.cnf

# ---------------------------------------------------------------------------
# Image Version
image_version: "P0S3-07-5-00"
# ---------------------------------------------------------------------------
# Proxy Adressen
proxy1_address: "192.168.0.107"
proxy2_address: "192.168.0.107"
# proxy3_address: "192.168.0.107"
# proxy4_address: "192.168.0.107"
# proxy5_address: "192.168.0.107"
# proxy6_address: "192.168.0.107"
#
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"

# Emergency Proxy info
proxy_emergency: "192.168.0.107"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "192.168.0.107"
proxy_backup_port: "5060"
 
# Outbound Proxy info
outbound_proxy: "192.168.0.107"
outbound_proxy_port: "5060"

# HTTP Proxy Support 
# http_proxy_addr: ""
# http_proxy_port: "80"
 
# ---------------------------------------------------------------------------
# NAT/Firewall Traversal
# nat_enable: "0"
# nat_address: ""
# voip_control_port: "5061"
# start_media_port: "16384"
# end_media_port:  "32766"
# nat_received_processing: "0"

# ---------------------------------------------------------------------------
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"
 
# Joerg, 080702 disabled
# Codec for media stream (g711ulaw (default), g711alaw, g729)
# preferred_codec: "none"
# Joerg, 080702 disabled
 
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"

# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"
 
# 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
# Default 500 msec
timer_t1: "500"               
# Default 4 sec
timer_t2: "4000"
# Default 11
sip_retx: "10"
# Default 7
sip_invite_retx: "6"
# Default 180 sec
timer_invite_expires: "180" 
 
# ---------------------------------------------------------------------------
# Setting for Message speeddial to UOne box
messages_uri: "*97"

# ---------------------------------------------------------------------------
# TFTP Phone Specific Configuration File Directory
# tftp_cfg_dir: "./"

# ---------------------------------------------------------------------------
# changed by JF
# 
# Time Server
sntp_mode: "unicast"
sntp_server: "de.pool.ntp.org"
# Summer time
# time_zone: "EET"
# Winter time
time_zone: "CET"
time_format_24hr: "1"
# date_format: "DD-MM-YY"
date_format: "D/M/Y"
# date_format: "D.M.Y"
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"
 
# ---------------------------------------------------------------------------
# Telephone Settings
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
# Default 0 (Do Not Disturb feature is off)
# dnd_control: "0"
 
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
# Default 0 (Disable sending all calls as anonymous)
# callerid_blocking: "0"
 
# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
# Default 0 (Disable blocking of anonymous calls)
# anonymous_call_block: "0"
 
# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1"

# Sync value of the phone used for remote reset  (Default 1)
# sync: "1"

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

# XML file that specifies the dialplan desired
dial_template: "dialplan"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"

# ARP
garp: "no"

# ---------------------------------------------------------------------------
# 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: "1"		; 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

# ---------------------------------------------------------------------------
# XML URLs
# directory_url: "http://192.168.0.107/cisco/vcdir.php"
logo_url: "http://192.168.0.107/cisco/bmp/IBM.bmp"	
services_url: "http://192.168.0.107/cisco/services/index_cisco.php"
directory_url: "http://192.168.0.107/cisco/services/PhoneDirectory.php"


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

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

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

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

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

# Telefonnummern automatisch vervollstaendigen - 0-Disabled, 1-Enabled (default)
autocomplete: "1"


# eof

Die Meldungen des Telefons und von Asterisk CLI
Das Telefon zeigt bei show status die Meldung:

Current Phone Status
--------------------
E640 REG msg unsupported: in 404, request failure
W310 1 Error(s) Parsing: SIPDefault.cnf

Ohne das ich telefoniere oder sonstwas, zeigt das Asterisk CLI bei sip set debug diese Meldung:

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK61424084;received=192.168.0.105
From: <sip:[email protected]>;tag=000785056314006403b5f60c-24d11735
To: sip:[email protected];tag=as7d14dadf
Call-ID: [email protected]
CSeq: 149 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

Bei eigehendem Anruf kommt im Asterisk CLI diese Nachricht:

<--- SIP read from 192.168.0.105:50570 --->
REGISTER sip:192.168.0.107 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK4a98b314
From: <sip:[email protected]>;tag=00078505631400c44f941218-1e5283e1
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 03 Jul 2008 08:36:29 GMT
CSeq: 197 REGISTER
User-Agent: Cisco-CP7960G/7.5
Contact: <sip:[email protected]:5060>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000785056314>"
Content-Length: 0
Expires: 3600


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.105 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK4a98b314;received=192.168.0.105
From: <sip:[email protected]>;tag=00078505631400c44f941218-1e5283e1
To: sip:[email protected];tag=as2a369fcd
Call-ID: [email protected]
CSeq: 197 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
 sterisk*CLI>
<--- SIP read from 192.168.0.105:50571 --->
REGISTER sip:192.168.0.107 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK1403a97b
From: <sip:[email protected]>;tag=00078505631400c55384ac95-1052e9fc
To: sip:[email protected]
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 03 Jul 2008 08:36:30 GMT
CSeq: 197 REGISTER
User-Agent: Cisco-CP7960G/7.5
Contact: <sip:[email protected]:5060>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000785056314>"
Content-Length: 0
Expires: 3600


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.105 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK1403a97b;received=192.168.0.105
From: <sip:[email protected]>;tag=00078505631400c55384ac95-1052e9fc
To: sip:[email protected];tag=as338afe87
Call-ID: [email protected]
CSeq: 197 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

Really destroying SIP dialog '[email protected]' Method: REGISTER
Really destroying SIP dialog '[email protected]' Method: REGISTER

Hat jemand eine Idee?

Gruss Horgee

-----------------
Trixbox 2.6 (13.06.08)
capi mit fritz card pci
Cisco 7960G , FW P0S3-07-5-00