I have been able to set up freepbx on a raspberry pi with no issue ia able to make internal calls from my pc and iPhone to test out extensions and it works great.
My issue is when i try and connect my Cisco 7940G phone to network
I am able to get the phone to connect with no issue but i cant dial or receive calls like i can with the other extensions
i have upgraded the phones to the latest SIP firmware
i am attaching my SIPDefault and my SIP00XX.cnf
I also have a asterisk log
SPIDefault.cnf
# Proxy Server
proxy1_address: " 192.168.1.19"
# Proxy Server Port (default - 5060)
proxy1_port: "5060"
# Emergency Proxy info
proxy_emergency: "192.168.1.19"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "192.168.1.19"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"
# 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: "2" ; 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: "*97"
# TFTP Phone Specific Configuration File Directory
#tftp_cfg_dir: "./"
# Time Server
sntp_mode: "unicast"
sntp_server: "us.pool.ntp.org"
time_zone: "EST"
dst_offset: "0"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
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: "0"
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
K0018B9A03F7B.cnf
line1_name: "304"
line1_shortname: "Office"
line1_displayname: "304"
line1_authname: "304"
line1_password: "Extpass203!"
This is my asterisk log - im also getting this error
[2018-02-20 02:53:32] ERROR[1098] pjproject: sip_transport. Error processing 1050 bytes packet from UDP 192.168.1.13:50740 : PJSIP syntax error exception when parsing 'To' header on line 4 col 14:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5061;branch=z9hG4bK1e5db2b9
From: "304" <sip:[email protected]>;tag=0018b9a03f7b000460e095c2-1b7b82a4
To: <sip:303@ 192.168.1.19>
Call-ID: [email protected]
Max-Forwards: 70
Date: Tue, 20 Feb 2018 07:53:28 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:[email protected]:5061;user=phone;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "304" <sip:304@ 192.168.1.19>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 276
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 28025 0 IN IP4 192.168.1.13
s=SIP Call
t=0 0
m=audio 29892 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
-- end of packet.