Endpoint Configuration doesnt work

mattrh

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#1
so here is whats going on. i log on to my local elastix server via the web. click pbx, then press Endpoint Configuration tab. i press scan (192.168.0.0/24) and it only finds 2 of my 30 phones. Whats going on? any ideas or what to look for????


thanks for the help
 

danardf

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#2
Yes.

Have you look each MAC address to compare the result with this list ?

If you haven't these MAC Address into this list, the extension can not be detected!
 

mattrh

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#3
Is there a way to add a MAC address to the list. Mine is not listed..

00:19:56:a6:ab:90

or can it programed by hand? and how?

Thanks for the help
 

danardf

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#4
Yes, it's possible.
Look at this post

Please, provide for increase this list on the Wiki.
you can edit this list for adding some Mac Address.

Like that, the Elastix team can use this list for the next versions.
You see? ;)
 

mattrh

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#5
well i have added my MAC for my cisco 7940. but i think i found the source of the problem. the tftp service might not be working. what are the codes to start tftp and stop? i did a port scan on my Elastix server and port 69 didnt respond or show as open.
 

danardf

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#6
Try to download this zip file and try it:

Make a directory backup, and replace / install the files.

Let me know.
 

mattrh

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#7
no luck, pick the config for the phone in the drop down list and the extension. and i get "Configured without incident"

00:19:56:A6:AB:90 192.168.0.86 Cisco / Unknown Configured without incident.



phone still isnt working or showing any extension....

:(
 

mattrh

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#8
ok the cisco phone was programed but the new issue is:

L1 and L2 do not work. i dont think they are connecting to the asterisk system. any ideas?
 

danardf

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#9
Maybe that yes...

Look at the config Cisco files:
SIP[mac].cnf
SIPDefault.cnf

Verifiy that you have the good IP address, secret code, account...
 

mattrh

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#10
here is my SIPDefault.cnf

# Image Version
image_version: "P0S3-08-6-00"

# Proxy Server Address
proxy1_address: "192.168.0.12"

# Proxy Server Port (default - 5060)
proxy1_port:"5060"

# Emergency Proxy info
proxy_emergency: "192.168.0.12"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "192.168.0.12"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: "192.168.0.12"
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: "g711ulaw"

# TOS bits in media stream [0-5] (Default - 5)
# tos_media: "5"
dscpForAudio: 184

# 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"

#Subdirectory config file location
#tftp_cfg_dir: /tftpboot/configs/sipphone
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"

# Time Server
sntp_mode: "unicast"
sntp_server: "192.168.0.12"
#time_zone: "PST"
#dst_offset: "1"
#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"

# URL for external Phone Services
services_url: "http://192.168.0.12/xmlservices/index.php"

# URL for external Directory location
directory_url: "http://192.168.0.12/xmlservices/E_book.php"

# URL for branding logo
logo_url: "http://192.168.0.12/images/bmp/asterisk-tux.bmp"

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




here is my SIP[mac].cnf


# Cisco SIP Configuration

phone_label: "Matt Herson"
line1_name: "4013"
line1_authname: "4013"
line1_shortname: "L1"
line1_displayname: "Matt Herson"
line1_password: "4013"
line2_name: "4013"
line2_authname: "4013"
line2_shortname: "L2"
line2_displayname: "Matt Herson"
line2_password: "4013"


it all looks correct.
 

danardf

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#11
Yes all is right.

What do you use into your LAN?
What's the result : sip show peers?
 

mattrh

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#12
i don't know what you are asking in your first question, but to answer your second one:

fapphonesvr*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
5001 (Unspecified) D N 0 UNKNOWN
4999 (Unspecified) D N 0 UNKNOWN
4998 (Unspecified) D N 0 UNKNOWN
4997 (Unspecified) D N 0 UNKNOWN
4996 (Unspecified) D N 0 UNKNOWN
4995 (Unspecified) D N 0 UNKNOWN
4994 (Unspecified) D N 0 UNKNOWN
4017 (Unspecified) D N 0 UNKNOWN
4016/4016 192.168.0.34 D N 5060 OK (11 ms)
4015/4015 192.168.0.21 D N 5060 OK (10 ms)
4014 (Unspecified) D N 0 UNKNOWN
4013/4013 (Unspecified) D N 0 UNKNOWN
4012/4012 (Unspecified) D N 0 UNKNOWN
4011/4011 (Unspecified) D N 0 UNKNOWN
4010/4010 192.168.0.25 D N 5060 OK (13 ms)
4009/4009 192.168.0.30 D N 5060 OK (10 ms)
4008/4008 192.168.0.29 D N 5060 OK (20 ms)
4007/4007 192.168.0.28 D N 5060 OK (10 ms)
4006/4006 192.168.0.27 D N 5060 OK (9 ms)
4005/4005 192.168.0.26 D N 5060 OK (10 ms)
4004/4004 192.168.0.31 D N 5060 OK (10 ms)
4003/4003 192.168.0.24 D N 5060 OK (11 ms)
4002/4002 192.168.0.33 D N 5060 OK (11 ms)
4001/4001 192.168.0.22 D N 5060 OK (10 ms)
4000/4000 192.168.0.23 D N 5060 OK (10 ms)
25 sip peers [Monitored: 13 online, 12 offline Unmonitored: 0 online, 0 offline]
 

mattrh

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#13
just a question, should the dial plan look like this:
dialplan.xml

<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/>
</DIALTEMPLATE>
 

danardf

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#14
Yes it's same configuration.

So... The Elastix server is 192.168.0.12: ok!
And the Cisco phone, what's the ip address?

Into the Cisco there is:
IP address.
subnet mask
gateway ?

It's a static IP or dynamic IP?

Maybe the router don't provide all the config (DNS, gateway...Etc)
 

danardf

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#15
What do you use into your LAN?
Router, Switch, VLAN, gateway.... etc
 

mattrh

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#16
sonicwall router (gateway) 192.168.0.1
subnet 255.255.255.0
that phone has a dynamic ip 192.168.0.86

all other phones are static and have all the same info

DNS
192.168.0.53 (windows sbs box)
 

danardf

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#17
Hmmm I forgot..
If the IP address is goog and the Cisco phone is connected to the LAN you can make a telnet cmd from your server like that:
Code:
# telnet 193.107.20.86
Trying 193.107.20.86...
Connected to SIP000F23E71C8A (193.107.20.86).
Escape character is '^]'.


Password :cisco

Cisco Systems, Inc. Copyright 2000-2005
Cisco IP phone  MAC: 000f:23e7:1c8a
Loadid:  SW: P0S3-08-8-00  ARM: PAS3ARM1  Boot: PC030301  DSP: 4.0(2.0)[A0]
SIP Phone>
 

mattrh

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#18
i was able to connect:



Password :*****

Cisco Systems, Inc. Copyright 2000-2005
Cisco IP phone MAC: 0019:56a6:ab90
Loadid: SW: P0S3-08-6-00 ARM: PAS3ARM1 Boot: PC030301 DSP: 4.0(2.0)[A0]
SIP Phone>
 

mattrh

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#19
do you have the new firmware P0S3-08-8-00. i don't have a cisco subscription so i can not download it.
 

danardf

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#20
Ok .. Now we know that the link is good, that the cisco is up and ready.
Only problem... the proxy.


I don't know why, I had make an install with 2 cisco 7960, and no problem.


Can you make a traceroute -p 5060 IP_Cisco
 

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