I have a similar, if not the same problem. I’m retiring an old server running Asterisk 1.4.11 (compiled myself, not installed as part of any package). I decided to give AsteriskNOW a try on the new server. I downloaded the ISO image (AsteriskNOW-3.0.0-x86_64-DVD.iso) and installed it, when I realized they use FreePBX as their GUI now, so I went with it…first time using FreePBX.
I created extension 201 to test my Cisco 7960 (SIP image) phone, but it will not register. I created extension 251 connected to it just fine with my WiFone app on my iPhone (using SIP).
I’ve verified that the Cisco 7960 is properly tftp’ing the OS79XX.TXT and the SIPXXX.cnf file just fine. FYI, the phone is running version P0S3-06-3-00 of the SIP image.
Watching tcpdump, I saw the tftp traffic and then the phone sending UDP to 5060 on the server without any replies. I turned on SIP debug and included the output at the bottom of this post. As you’ll see, the phone (192.168.23.252) is sending REGISTER packets and the server (192.168.23.24) says nothing about it.
I decided I’d see if this was an Asterisk issue or FreePBX issue, so wiped the server clean and put CentOS 6.4 on it, then I downloaded certified-asterisk-11.2-current.tar.gz from Digium and installed it (and it’s prerequisite packages). I used the same config files from my Asterisk 1.4.11 server (copied them verbatim from the tftpboot dir and the /etc/asterisk dir). Conclusion is that I had the same exact problem. SIP Debug shows register requests and nothing else.
This was a working setup with 1.4.11 and I’m quite sure that I can turn the old server back on and put it back in as 192.168.23.24 on the network and get it working again. I’m thinking there’s got to be something that Digium did between 1.4.11 and 11.2 that either changes the necessary config for the 7960 phone, or just breaks it altogether. I’ve not been able to find any suggested solutions online so I decided to post here in case someone might know something. I’d appreciate any feedback and would be glad to post additional details if needed. Here’s that SIP debug, followed by the contents of SIPDefault.cnf and SIPXXX.cnf:
<— SIP read from UDP:192.168.23.252:50377 —>
REGISTER sip:192.168.23.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.252:5060;branch=z9hG4bK21ecf965
From: sip:[email protected];user=phone
To: sip:[email protected];user=phone
Call-ID: [email protected]
Date: Sat, 19 Oct 2013 22:24:00 GMT
CSeq: 393 REGISTER
User-Agent: CSCO/6
Contact: sip:[email protected]:5060;user=phone
Content-Length: 0
Expires: 120
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:192.168.23.252:50377 —>
REGISTER sip:192.168.23.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.252:5060;branch=z9hG4bK21ecf965
From: sip:[email protected];user=phone
To: sip:[email protected];user=phone
Call-ID: [email protected]
Date: Sat, 19 Oct 2013 22:24:00 GMT
CSeq: 393 REGISTER
User-Agent: CSCO/6
Contact: sip:[email protected]:5060;user=phone
Content-Length: 0
Expires: 120
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:192.168.23.252:50377 —>
REGISTER sip:192.168.23.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.252:5060;branch=z9hG4bK21ecf965
From: sip:[email protected];user=phone
To: sip:[email protected];user=phone
Call-ID: [email protected]
Date: Sat, 19 Oct 2013 22:24:00 GMT
CSeq: 393 REGISTER
User-Agent: CSCO/6
Contact: sip:[email protected]:5060;user=phone
Content-Length: 0
Expires: 120
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:192.168.23.252:50377 —>
REGISTER sip:192.168.23.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.252:5060;branch=z9hG4bK21ecf965
From: sip:[email protected];user=phone
To: sip:[email protected];user=phone
Call-ID: [email protected]
Date: Sat, 19 Oct 2013 22:24:00 GMT
CSeq: 393 REGISTER
User-Agent: CSCO/6
Contact: sip:[email protected]:5060;user=phone
Content-Length: 0
Expires: 120
<------------->
— (11 headers 0 lines) —
bishopCLI>
bishopCLI>
<— SIP read from UDP:192.168.23.252:50377 —>
REGISTER sip:192.168.23.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.252:5060;branch=z9hG4bK21ecf965
From: sip:[email protected];user=phone
To: sip:[email protected];user=phone
Call-ID: [email protected]
Date: Sat, 19 Oct 2013 22:24:00 GMT
CSeq: 393 REGISTER
User-Agent: CSCO/6
Contact: sip:[email protected]:5060;user=phone
Content-Length: 0
Expires: 120
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:192.168.23.252:50377 —>
REGISTER sip:192.168.23.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.23.252:5060;branch=z9hG4bK21ecf965
From: sip:[email protected];user=phone
To: sip:[email protected];user=phone
Call-ID: [email protected]
Date: Sat, 19 Oct 2013 22:24:00 GMT
CSeq: 393 REGISTER
User-Agent: CSCO/6
Contact: sip:[email protected]:5060;user=phone
Content-Length: 0
Expires: 120
SIPDefault.cnf
Image Version
image_version: “P0S3-06-3-00”
Proxy Server Port
proxy1_port:“5060”
NAT/Firewall Traversal
nat_enable: “0”
Proxy Registration (0-disable (default), 1-enable)
proxy_register: “1”
Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: “120”
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: “0” ; 0-Disabled (default), 1-Enabled, 2-Privileged
Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: “0”
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: “2500”
#********* Release 2 new config parameters **********
TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: “./sipphones/”
Time Server
sntp_mode: "directedbroadcast"
sntp_server: "192.168.23.25"
time_zone: "CST"
dst_offset: "1"
dst_start_month: "March"
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”
####### New Parameters added in Release 4.0 #######
HTTP Proxy Support
http_proxy_addr: “” ; Address of HTTP Proxy server
http_proxy_port: 8050 ; 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: “192.168.23.25” ; restricted to dotted IP
The dynamic tftp server should be set to whatever your TFTP server is. This way, it
keeps the tftp server setting even though you might be using DHCP (default behavior
is to use the DHCP server as a tftp server, which is rarely correct.)
Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
SIPXXXXX.cnf file
SIP Configuration Generic File (start)
Proxy Server
proxy1_address: "192.168.23.24"
proxy2_address: "192.168.23.24"
proxy3_address: "192.168.23.24"
proxy4_address: "192.168.23.24"
proxy5_address: "192.168.23.24"
proxy6_address: “192.168.23.24”
Line 1 Settings
line1_name: “201” ; Line 1 Extension\User ID
line1_displayname: “MyDisplayName” ; Line 1 Display Name
line1_authname: “201” ; Line 1 Registration Authentication
line1_password: “password” ; Line 1 Registration Password
line2_name: “” ; Line 3 Extension\User ID
line2_displayname: “” ; Line 3 Display Name
line2_authname: “UNPROVISIONED” ; Line 3 Registration Authentication
line2_password: “UNPROVISIONED” ; Line 3 Registration Password
Line 3 Settings
line3_name: “” ; Line 3 Extension\User ID
line3_displayname: “” ; Line 3 Display Name
line3_authname: “UNPROVISIONED” ; Line 3 Registration Authentication
line3_password: “UNPROVISIONED” ; Line 3 Registration Password
Line 4 Settings
line4_name: “” ; Line 4 Extension\User ID
line4_displayname: “” ; Line 4 Display Name
line4_authname: “UNPROVISIONED” ; Line 4 Registration Authentication
line4_password: “UNPROVISIONED” ; Line 4 Registration Password
Line 5 Settings
line5_name: “” ; Line 5 Extension\User ID
line5_displayname: “” ; Line 5 Display Name
line5_authname: “UNPROVISIONED” ; Line 5 Registration Authentication
line5_password: “UNPROVISIONED” ; Line 5 Registration Password
Line 6 Settings
#line6_name: “” ; Line 6 Extension\User ID
#line6_displayname: “” ; Line 6 Display Name
#line6_authname: “UNPROVISIONED” ; Line 6 Registration Authentication
#line6_password: “UNPROVISIONED” ; Line 6 Registration Password
Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: “5060”
Backup Proxy info
proxy_backup: ""
proxy_backup_port: “5060”
Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: “5060”
NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: “0”
Phone Label (Text desired to be displayed in upper right corner)
phone_label: “MyPhone” ; Has no effect on SIP messaging
Time Zone phone will reside in
time_zone: CST
Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: “2” ; 0-Disabled (default), 1-Enabled, 2-Privileged
Phone prompt/password for telnet/console session
phone_prompt: “prompt” ; Telnet/Console Prompt
phone_password: “password” ; Telnet/Console Password
Enable_VAD (1-enabled, 0-disabled)
enable_vad: “0”
Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: phone
URL for external Directory location
logo_url: “http://192.168.23.24/asterisk/asterisk-tux.bmp” ; URL for branding logo to be used on phone display
DTMF Stuff
dtmf_outofband: "avt_always"
dtmf_inband: “0”
SIP Configuration Generic File (stop)
