Cisco 7940 unable to register

Kijush Maharjan · ‎01-23-2014

Hi,

I have a weird problem on cisco 7940 that is not registering to outside world. My setup is as below:

1. Elastix server on private IP.

2. All cisco phones connect with elastix server and gets registered with internal extensions.

3. While assigning sip account on line 2 it's not getting registered. I have configured particular SIPXXXXXXX.cnf file with proxy address and nat settings but still it's not registering.

Could anyone help me solve this issue as i might have missed some configs here?

Thanks,

Kijush

Manish Prasad · ‎01-27-2014

Hi Kijush,

Try to capture SIP packets and see what Elastix server is returning when 7940 sends registration request.

Rate all the helpful post.

Thanks

Manish

Kijush Maharjan · ‎01-27-2014

Hi manish,

Thanks for your reply.

Cisco 7940 doesn't have any problem with sip extensions from elastix as both exists on same subnet. But whenever i configure Sip account from another voip provider, registration fails. I configured nat and outbound proxy of voip provider and it starts working but then line 1 which is internal extension fails registration.

So is there any way to configure cisco phones to use two accounts from different provider?

Regards,

Kijush

Manish Prasad · ‎01-27-2014

Hi

You cant register to two different server at a time , one will always remain unavailable as they work in as primary and secondary .

Sent from Cisco Technical Support iPhone App

Kijush Maharjan · ‎01-27-2014

Hi,

Oh so that means I can't use both lines simultaneously for two different providers.

But i can use two different sip account with same sip server.

I am confused why cisco introduced dual line sip phones then...

Manish Prasad · ‎01-27-2014

Hi Kijush,

You can use both line for a single service provider. The purpose of creating extra lines is not to get it registered to different - different TFTP servers but to receive or call multiple numbers simultaneously.

IP phone register with TFTP server on the basis of MAC address/IP address not on the basis of lines.So in your case both TFTP servers will not be active at a time.

Rate all the helpful post.

Thanks

Manish

Kijush Maharjan · ‎01-28-2014

Hi Manish,

Thanks for your detailed information. Its more clear now.

Cheers.

Kijush Maharjan · ‎01-28-2014

Hi Manish,

I think your information is wrong. I have configured Outbound proxy to blank and both the SIP accounts are registering. We can parse the cnf file by our own and can modify however we want so it doesn't depend upon the automated system from TFTP.

Manish Prasad · ‎01-28-2014

Hi Kijush,

I would love to be wrong and it would be something new learn for me.

Now if i understand your scenario properly this is how its working ...

7940 - Line 1 - 2222 - Registered with xxx server

7940 - Line 2 - 3333 - Registered with yyy server

And you can call from both these lines ?

Can you please share the cnf file (removing secure IP/s)

Thanks

Manish

Kijush Maharjan · ‎01-28-2014

Hi Manish,

Yes you are absolutely right.

SIPDefault.cnf:

[root@pbx tftpboot]# cat SIPDefault.cnf

# Image Version

image_version: "P0S3-8-12-00"

# Proxy Server Address

proxy1_address: "10.0.2.11"

# Proxy Server Port (default - 5060)

proxy1_port:"5060"

# Emergency Proxy info

proxy_emergency: "10.0.2.11"

proxy_emergency_port: "5060"

# Backup Proxy info

proxy_backup: "10.0.2.11"

proxy_backup_port: "5060"

# Outbound Proxy info

outbound_proxy: "0.0.0.0"

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"

# 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: "g711alaw"

# 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: "10.0.2.11"

time_zone: "GMT"

#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://10.0.2.11/bmp/index.php"

# URL for external Directory location

directory_url: "http://10.0.2.11/bmp/edir.php"

# URL for branding logo

logo_url: "http://10.0.2.11/bmp/elastix.bmp"

# Remote Party ID

remote_party_id: 1 ; 0-Disabled (default), 1-Enabled

[root@pbx tftpboot]#

SIPXXXXXX.cnf

[root@pbx tftpboot]# cat *ABDD.cnf

# Cisco SIP Configuration

phone_label: "Kijush Maharjan"

line1_name: "285"

line1_authname: "285"

line1_shortname: "285-L1"

line1_displayname: "Kijush Maharjan"

line1_password: "********"

line2_name: "kijush"

line2_authname: "kijush"

line2_shortname: "kijush-L2"

line2_displayname: "Kijush Maharjan"

line2_password: "********"

proxy2_address: "192.168.1.1"

proxy2_port: "5060"

nat_enable: "1"

preferred_codec: "g729a"