Cisco 7960 P003-08-8-00 and Trixbox 2.2.5 CE

adm
Posts: 3
Member Since:
2007-02-02

Need help on this. I am adding a new 7960 phone which is setup with the latest Firmware on my trixbox 2.2.5 ce. The trixbox and the 7960 are on the same subnet, behind my internet router. Trixbox is working fine with othet ATA (PAP2 and RTP31).

The cisco phone is able to make successful outgoing calls to any of my other extensions,
The problem is the Cisco 7960 does not seem to register in the PBX to receive incoming calls. I have played a lot with the configuration and tried all options I know. I am posting below the current setup which allows outgoing, but not incoming to the cisco 7960.

The Cisco phone and the extension 304 in trixbox are setup as NAT. Looking at "show peers" on asterisk, the Cisco does not show up as registered (304 (Unspecified) D N 0 Unmonitored). If I call the extension 304 (cisco phone), asterisk debug says s-CHANUNAVAIL. If I change the "host" from dynamic to the cisco IP (extensions file) and then show peers, I see the cisco phone there. Now I try to call the Cisco phone (304), asterisk says the extension is ringing, but the nothing happens on the cisco phone.

Any ideas?

Thanks,
Marcello

<<<<<<<<<<<< SIPDefault.Cnf >>>>>>>>>>>>>>>>>>>

# SIP Default Generic Configuration File
image_version: P0S3-08-8-00
proxy1_address: "192.168.3.112" ; Can be dotted IP or FQDN
proxy_register: 1
timer_register_expires: 3600
preferred_codec: g711ulaw
dtmf_inband: 1
dtmf_outofband: avt
dtmf_db_level: 3
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
dial_template: dialplan
tftp_cfg_dir: "" ; Example: ./sip_phone/
sntp_server: "192.168.3.112" ; SNTP Server IP Address
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EAST ; 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: 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: 0 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format : D/M/Y
dnd_control: 0 ; Default 0 (Do Not Disturb feature is off)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)
dtmf_avt_payload: 101 ; Default 101
sync: 1 ; Default 1
proxy_backup: "192.168.3.112" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
proxy_emergency: "192.168.3.112" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)
enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable
# NAT/Firewall Traversal
nat_enable: 1 ; 0-Disabled (default), 1-Enabled
nat_address: "192.168.3.112" ; 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: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 32766 ; End RTP range for media (default - 32766)
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: "192.168.3.112" ; restricte

<<<<<<<<<<<< SIPDefault.cnf >>>>>>>>>>END>>>>>>>>>

SIP.cnf
<<<<<<<<<<<< SIP.cnf >>>>>>>>>>>>>>>>>>>
; phone-specific configuration file sample
line1_name : office
line1_authname : 304
line1_password : 3041234
l
####### 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: "Office" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "Office"
####### 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: "123" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none

<<<<<<<<<<<< SIP.cnf >>>>>>>>>>END>>>>>>>>>

Trixbox extension 304

[304]
type=friend
secret=3041234
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
pickupgroup=
nat=no
mailbox=304@device
host=dynamic
dtmfmode=rfc2833
disallow=all
dial=SIP/304
context=from-internal
canreinvite=yes
callgroup=
callerid=device <304>
allow=g729,ulaw
accountcode=



skykingoh
Posts: 1012
Member Since:
2007-12-17
Try NAT = No in the trix

Try NAT = No in the trix extension.

Here is my NAT section from SIPdefault.cnf

# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"

Scott



adm
Posts: 3
Member Since:
2007-02-02
I did that already and same

I did that already and same end results. Able to place calls, but no incoming. when you run "sip show peers" do you see the phone registered in asterisk?

thanks
Marcello



skykingoh
Posts: 1012
Member Since:
2007-12-17
Quote: do you see the phone
Quote:
do you see the phone registered in asterisk?

Yep, this is my test box....Extension 206 is the local Cisco phone.

trix01*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
DID-from-BV                64..x.x.x       N      5060     Unmonitored
Broadvox                   64.x.x.x        N      5060     OK (55 ms)
299                        (Unspecified)    D   N      0        UNKNOWN
291/291                    204.x.x.x     D   N      5060     OK (103 ms)
290/290                    207.x.x.x   D   N      1026     OK (77 ms)
280                        (Unspecified)    D   N      0        UNKNOWN
277                        (Unspecified)    D   N      0        UNKNOWN
270/270                    (Unspecified)    D   N      0        UNKNOWN
269/269                    (Unspecified)    D   N      0        UNKNOWN
222                        (Unspecified)    D   N      0        UNKNOWN
215                        (Unspecified)    D   N      0        UNKNOWN
214                        (Unspecified)    D   N      0        UNKNOWN
213                        (Unspecified)    D   N      0        UNKNOWN
210                        (Unspecified)    D   N      0        UNKNOWN
209                        (Unspecified)    D   N      0        UNKNOWN
208                        (Unspecified)    D          0        UNKNOWN
207                        (Unspecified)    D          0        Unmonitored
206/206                    192.168.0.148    D          5060     Unmonitored
205/205                    76.188.x.x   D   N      51912    OK (96 ms)
204/204                    75.187.x.x   D   N      5060     OK (185 ms)
203                        (Unspecified)    D          0        UNKNOWN
202/202                    (Unspecified)    D   N      0        UNKNOWN
201                        (Unspecified)    D          0        UNKNOWN
200                        (Unspecified)    D          0        UNKNOWN
24 sip peers [8 online , 16 offline]



igorek600
Posts: 63
Member Since:
2007-07-21
check dnd settings on a

check dnd settings on a phone and in freepbx



adm
Posts: 3
Member Since:
2007-02-02
Got it working now. I

Got it working now. I changed the line1_name to the extension # and add line1_shortname as well.

New sip.cnf.
; phone-specific configuration file sample
line1_name : 203
line1_shortname : 203
line1_authname : 203
line1_password : 1234

Thanks for the help.
Marcello



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