Cisco 7940G Newbie help -

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 cant include my SIPDefault, SIP00XX.cnf or my asterisk log - how can i do that says new users cant post links

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:[email protected] 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:[email protected] 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.

You have downgraded the phone from SCCP to SIP. The SIP image has a lot of problems because the phone just doesn’t have the computing horsepower to do a real, full SIP implementation.

The typical problem here is that the password is too long for the phone. Twelve characters is a pretty common “OK” length, but without looking at the server logs, there’s no way to know.

You need to check the /var/log/asterisk/full log file to see what issue you are actually having. There are several things in your configuration that look hinky. For example, your preferred codec is “none”, which I’m pretty sure is wrong - you need a codec.

Check out the log files and let us know what problem the server is seeing - we can help you from there.

Hi ,

I added a codec to the config and also made the password the same as the extension - its just for testing inhouse so thats fine with me for now.  Thanks !!! Gio 

here is my SIPXXXX.cnf

line1_name: "302"
line1_shortname: "Office"
line1_displayname: "302"
line1_authname: "302"
line1_password: "302"
preferred_codec: g711ulaw


here is what happens when i tail the log file and try to make a call internally ( i notice that there is a space after the @ symbol in my to address. )

/var/log/asterisk/full

[2018-02-21 21:56:06] ERROR[1098] pjproject:    sip_transport. Error processing 1050 bytes packet from UDP 192.168.1.13:50773 : 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=z9hG4bK64c1063d
From: "302" <sip:[email protected]>;tag=0018b9a03f7b0008098d3d81-3d981b1f
To: <sip:[email protected] 192.168.1.19>
Call-ID: [email protected]
Max-Forwards: 70
Date: Thu, 22 Feb 2018 02:55:58 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: "302" <sip:[email protected] 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 12346 0 IN IP4 192.168.1.13
s=SIP Call
t=0 0
m=audio 26616 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.

Good to hear from a fellow traveller…I am playing around with Cisco7960G and rasPBX as well. The good news is that Cisco79XX phones do work. I have had long passwords (longer than what you have) on these phones running SIP without any issues.

Probably you have got this right but anyway make sure that your proxy port is 5160 not 5060 if you are using chan_sip definitions for the handsets as the default for chan_sip is 5160 and pjsip is 5060 unless you have already changed them on Asterisk SIP settings. Does the phone show that it has registered with rasPBX?

Which version of SIP are you running on your phones? I am running pos3-8-12-00 and I read somewhere that SIP is buggy with later versions.

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