Cisco 7940 - 2 Lines, 2 Providers?

Hi,

I have a Cisco 7940 which I have recently set-up. It is Sip configured.

I have Line 1 set-up with VoipDiscount for outbound calls.

I would like to set-up Line 2 with another provider for inbound calls and outbound calls as a redundancy in the event of problems with VoipDiscount.

My problem is that Line 2 does not seem to connect to the provider (icon has a x by it). I also note that the outbound proxy setting seems to be universal and not line specific therefore I cannot seem to set a separate outbound proxy for Line 2.

Has anybody had any experience of this before? Is what I am trying to achieve possible?

Thank you very much for your assistance.

Matt

Hi Matt and welcome to Voipuser

Quote:

Is what I am trying to achieve possible?

Yes - and it should work. You either have a problem with the second account or an entry incorrect somewhere, just go back through it carefully and check again. Ping the server you are trying to connect to and make sure the address is right + double check your password. If you can post your settings (without passwords) we can take a look. Another option would be to test again with Voipuser as your second provider.

Hi,

I am bumping this because I made some progress last year but I am still having a problem with inbound calls.

The problem seems to be that if I set the proxy for voipdiscount then the voipon inbound will not function. If I change the proxy to voipon then the voipdiscount outbound will not work.

Is this a problem with the 7940 or is there anyway around this? I can't seem to see a way of setting a proxy for each line separately.

Thank you very much

Matt

Can you let us know which SIP image version you are using?

(On the phone press - Settings»Status»Firmware Version) AND as per above...

Quote:

If you can post your settings (without passwords) we can take a look

Hi,

Thank you for your reply

I think these are the right files please let me know if not.

Thanks again

Matt

# Image Version
image_version: "P0S3-08-3-00"

# Proxy Server
proxy1_address: "sip.voipdiscount.com" # IP address here alternatively
proxy2_address: "sip.voipon.co.uk" # IP address here alternatively

# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:"5060"

# Emergency Proxy info
proxy_emergency: "" # IP address here alternatively
proxy_emergency_port: "5060"

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

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5065"

# NAT/Firewall Traversal
nat_enable: "1"
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: "1" ; 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" ~np~# 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: "**Removed**" # IP address here alternatively
time_zone: "GMT"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
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: "" # IP address here alternatively

# URL for external Directory location
directory_url: "" # IP address here alternatively

# URL for branding logo
logo_url: "" # IP address here alternatively

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

# SIP Configuration Generic File

# Image Version
image_version: P0S3-08-3-00
phone_label: "**Removed**"

# Line 1 appearance
line1_displayname: "VoipDiscount"
line1_shortname:"**Login**"
line1_name: **Login**
line1_authname: "**Login**"
line1_password: "**Password**"

# Line 2 appearance
line2_displayname: "Voipon"
line2_shortname: "**Login**"
line2_name: **Login**
line2_authname: "**Login**"
line2_password: "**Password**"

# Line 3 appearance
line3_displayname: ""
line3_shortname: ""
line3_name: UNPROVISIONED
line3_authname: "UNPROVISIONED"
line3_password: "UNPROVISIONED"

# Line 4 appearance
line4_displayname: ""
line4_shortname: ""
line4_name: UNPROVISIONED
line4_authname: "UNPROVISIONED"
line4_password: "UNPROVISIONED"

# Line 5 appearance
line5_displayname: ""
line5_shortname: ""
line5_name: UNPROVISIONED
line5_authname: "UNPROVISIONED"
line5_password: "UNPROVISIONED"

# Line 6 appearance
line6_displayname: ""
line6_shortname: ""
line6_name: UNPROVISIONED
line6_authname: "UNPROVISIONED"
line6_password: "UNPROVISIONED"

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone)

# Phone Password (Password to be used for console or telnet login)
phone_password: "**Removed**" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none

Hi,

Please can someone assist me with this? I should say the above are the settings that I have working currently. I have tried alternative settings to get both lines to dial out but it seems I can only dial out from one or the other if I do that.

Thank you very much

Matt