Please specify the group
Sip ρυθμίσεις του πελάτη
Αν η SIP client σας είναι πίσω από ένα τείχος προστασίας, βεβαιωθείτε ότι το τείχος προστασίας σας δεν μπλοκάρει τη θύρα UDP 5060 και το εύρος των θυρών UDP που χρησιμοποιούνται για τη μετάδοση φωνής (βλ. firewall σας και SIP client τεκμηρίωση για λεπτομέρειες)
Εδώ είναι τα παραδείγματα των διαφορετικών πελατών διαμόρφωσης VoIP (χάρη στους χρήστες μας!)
For
open vpn please check
Configuration X-lite
Download the X lite.
Click in the Sip Account Settings.
and give your Display name and the User name Authorization user, password
and the Domain

To download the
x-lite click
Configuration for Eyebeam
Please check what is more good for you to start make calls with our system.
Configuration Ekiga soft phone with the CIP Telecom
Its easy to setup start to sell voip business with us.

If you like to download
please visit
Configuration Twinkle
Configuration PCBest ActiveX
Configuration Zoiper
You can setup IAX or SIP with us please only check ho server you work
Our Server in USA is 12.47.45.239
Our Server in Europe is 213.175.221.14
Configuration Fring
On Add-on selection screen select SIP
Select SIP service name screen - select Other
User-ID: username
Password: password
Proxy Address: 213.175.221.14
You're all set.
Configuration Express Talk

If you like to download
the Express Talk
Configuration Nokia E51
Here are the settings that work on my Nokia E51 phone and should be similar on other Nokia phones.
Go to Menu-> Tools-> Settings-> Connection-> SIP Settings-> New Sip
profile (add a new profile as below)
Profile name: CIP Telecom (or whatever you want)
Service profile: IETF
Default access point: your wifi access point
Public user name: 123456789@sip.213.175.221.14 (where the numbers
correspond to your CIP Telecom user name)
use compression: No
Registration: Always on
Use security: No
Registrar server
Registrar server address: sip:213.175.221.14
Realm: sip.callwithus.com
Username:123456789 (CIP Telecom user name)
Password: your password (CIP Telecom 6 digit password)
Transport type: UDP
Port: 5060
Configuration Nokia 6300i
To give the Nokia 6300i the sip setup it must give the Nokia6300i.xml.
Click to
download
Configuration Zyxel P2000W
Configuration Zyxel MAX-206M2
Siemens Gigaset S450
Configuration UTStarcom F1000
Configuration Polycom Soundpoint 330
Configuration Grandstream GXP200
Build
Configuration Grandstream HT 286
Configuration Grandstream HT-496
Configuration Telco Systems AC-211
Configuration Innomedia MTA6328 -2Re
SunRocket closed its doors a while ago, and for those that still their Innomedia devices:
Only for those who are reedy to wipe out the config and use it for other VOIP providers:
1. Disconnect to WAN (no internet)
2. Connect the gizmo via LAN to computer and login to it using user id
http://192.168.251.1
id=user, pass=welcome
3. Cut and past the following link and enter to Restore factory default
http://192.168.251.1/restore2.ssi
4. Power cycle the gizmo and login to admin link with factory default password
http://192.168.251.1/Voip_adminPage.htm
id=admin, pass=slapshot (for V3.0.77, V3.0.75)
5. Disable Provisioning
Menu -- IP Network -- Provisioning Setting
Remaining steps... see below pics.
Pic #1 - Version information
Pic #2 - SIP Setting
Pic #3 - User Connection /Account Information
Photo 1

Photo 2

Photo 3
Configuration Zoom 5801
Configuration VTech IP8100
Configuration Thomson ST780 WL
Configuration FRITZ Box 7140
Configuration FRITZ Box 7270
Build
Configuration Aastra 51i
You can fix your Aasrta for your office or your business.

You can visit the web site of the
Aastra click.
Configuration DrayTek Vigor 2700V with the CIP Telecom
Configuration TrixBox
Example of SIP trunk setup:
Outbound Caller ID:
Never Override CallerID: checked
Maxium channels: 10
Dial Rules:
Outbound Dial Prefix:
Trunk Name: CIP Telecom
PEER Details:
context=from-trunk
host=sip.cip-tele.com
qualify=no
username=username
secret=password
type=friend
insecure=invite
USER Context:
USER Details:
Dial command (extensions.conf):
_X. => 1,Dial(SIP/CIPTelecom/${EXTEN},120)
_*X. => 1,Dial(SIP/CIPTelecom/${EXTEN},120)
If you defined a DID in trixbox configuration as 1234567890, then set the DID destination to
SIP/username/1234567890.
Configuration Linksys SPA 9000 PBX
Configuration Linksys PAP2
Configuration Linksys pap2t
Build
Configuration Sipura SPA-2000
Sipura Linksys SPA-3000
Configuration Linksys SPA 2102
Configuration Linksys WIP 300
WIP300 configuration
Network Profile
NAT STUN
STUN Address
STUN Port 3478
SIP account settings
Phone Number username
Auth ID username
Auth Password password
SIP Domain 213.175.221.14
Proxy Address 213.175.221.14
Advanced settings
Reg. Timer(sec) 120
Codec AUTO
Pkt Time(ms) Default
Outband DTMF ON
Configuration Cisco ATA 186
Configuration Cisco 7960
Build your own
Original Template found at
http://www.voip-info.org/wiki/view/cisco+mass+deployment
If you are using a NAT proxy please make sure to forward UDP ports 5060/5060 and 16384/32768Β (Start port/End Port) to your internal IP number for your phone you might want to setup a static IP or have your router assign one for the Phone.Β Along with other config files mentioned at " http://www.voip-info.org/wiki/view/cisco+mass+deployment" include these two files in your tftp directory for your phone, "SIPDefault.cnf" and "SIPXXXXXXXXX.cnf" XX represent your device's Mac Address.Β I am also attaching some of the configuration files
FILENAME: SIPDefault.cnf
CONTENTS:
# Image Version
image_version: "P0S3-08-3-00" # Input your specific Firmware version here
# Proxy Server
proxy1_address: " 213.175.221.14" # IP address here alternatively
# Proxy Server Port (default - 5060)
proxy1_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: "5060"
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "XX.XXX.XXX.XX" # Your external IP adress
voip_control_port: "5060"
start_media_port: "16384"
end_media_port: "32768"
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"
# 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" ~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: "sip.callwithus.com" # IP address here alternatively
time_zone: " EST" # Set to your timezone
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 branding logo
logo_url: " http://pbx.mycompany.com/cisco/logo.bmp"
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
FILENAME: SIPXXXXXXXXX.cnf Where the XXs are the MAC address of your Cisco 7960
CONTENTS:
# SIP Configuration Generic File
# Image Version
image_version: P0S3-08-3-00
phone_label: " "
# Line 1 appearance
line1_displayname: "username"
line1_shortname:"CW"
line1_name: username
line1_authname: "username"
line1_password: "password"
# Line 2 appearance
line2_displayname: ""
line2_shortname: ""
line2_name: UNPROVISIONED
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
# 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: "cisco" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
dialplan.xml:
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>
XMLDefault.cnf.xml:
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model="IP Phone 7940">P003-07-4-00</loadInformation8>
<loadInformation7 model="IP Phone 7960">P003-08-3-00</loadInformation7>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
Configuration Cisco 79x1
Build your own
Also works with 7941G.
These phones are very different than the 79x0 series, but I have one successfully working with CIP Telecom.
Support added "nat=no" to your SIP proxy, which enables the attached configuration to load.
I have the following ports forwarded to the phone's internal IP address: UDP/5060, UDP/16384-16393
Replace the following stings in the
file:
%INTERNET_HOSTNAME_OR_IP% (can use dynamic DNS or public IP address for NAT router on user side) %SHORTNAME% (label appears next to line button on phone's display)
%CWS_ACCOUNT_NUM% (CIP Tlecom account number, note that this appears 3 times for each configured line - name, authName, and contact elements)
%CWS_ACCOUNT_PASSWORD% CIP Telecom account password)
The file name must be of format "SEP" + phone MAC address in hex + ".cnf.xml" and be ANSI/ASCII (not UTF) encoded.
The following entries on the VOIP Wiki are also good resources for configuring Cisco 79x1 phones without using the Cisco CallManager PBX:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
http://www.voip-info.org/wiki/index.php?page=Standalone%20Cisco%207941/7961%20without%20a%20local%20PBX
Configuration Cisco Call Manager
dial-peer voice 77 voip
destination-pattern 77
session protocol sipv2
session target ipv4:213.175.221.14
dtmf-relay rtp-nte
codec g711ulaw
clid network-number username
!
sip-ua
authentication username username password password
nat symmetric check-media-src
registrar ipv4:213.175.221.14 expires 3600
sip-server ipv4:213.175.221.14.
Configuration Cisco CME
###########################################
# IMPORTANT: If behind Cisco PIX firewall, must have sip fixup enabled on firewall else
# Sip debug output symptom will show following:
#
# \"SIP/2.0 400 Bad Request - \'Invalid IP Address\'
# \"Invalid URL in incoming INVITE\"
#
# If using Cisco ASA firewall, remove \"inspect sip\" from \"class inspection_default\" else
# Symptom will be one-way audio to mobile (cellular) phones only...
#
############################################
boot-start-marker
boot system flash c2800nm-adventerprisek9-mz.124-24.T1.bin
boot-end-marker
!
clock timezone PST -8
clock summer-time PDT recurring
!
!
ip domain name doda.us
ip host cme.doda.us 192.168.0.38
ip name-server 208.67.222.222 4.2.2.3
no ipv6 cef
ntp master
ntp server 0.north-america.pool.ntp.org prefer
ntp server 1.north-america.pool.ntp.org
ntp server 2.north-america.pool.ntp.org
ntp server 3.north-america.pool.ntp.org
!
voice service voip
no notify redirect ip2ip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
sip
header-passing
registrar server expires max 120 min 60
no update-callerid
no call service stop
!
voice class codec 1
codec preference 1 g711ulaw
!
!
voice translation-rule 1
rule 1 /13104697570\\(\\)/ /5000\\1/
!
voice translation-rule 2
rule 1 /.*\\(....\\)$/ /\\1/
!
voice translation-rule 4
rule 1 /.*\\(\\)/ /3104697570\\1/
!
voice translation-rule 5
rule 1 /.*\\(..........\\)/ /1\\1/
!
voice translation-rule 6
rule 1 /.*\\(\\)/ /13104697570\\1/
!
!
voice translation-profile CLID_INTL
translate calling 6
!
voice translation-profile FROM_CIP-TELECOM
translate called 1
!
voice translation-profile TO_CIP-TELECOM
translate calling 4
translate called 5
!
voice translation-profile TO_CUE_VM
translate called 2
translate redirect-called 2
!
!
voice-card 0
!
!
username admin privilege 15 password *****
username xadmin privilege 15 password *****
username perryb privilege 15 password *****
!
interface FastEthernet0/0
ip address 192.168.0.38 255.255.255.0
duplex auto
speed auto
!
interface Service-Engine0/0
ip unnumbered FastEthernet0/0
service-module ip address 192.168.0.22 255.255.255.0
service-module ip default-gateway 192.168.0.38
!
ip route 0.0.0.0 0.0.0.0 192.168.0.1
ip route 192.168.0.22 255.255.255.255 Service-Engine0/0
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
tftp-server flash:RingList.xml
tftp-server flash:DistinctiveRingList.xml
tftp-server flash:klaxons.raw
tftp-server flash:flintphone.raw
tftp-server flash:Vibe.raw
tftp-server flash:P00308000500.bin
tftp-server flash:P00308000500.loads
tftp-server flash:P00308000500.sb2
tftp-server flash:P00308000500.sbn
!
voice-port 0/1/0
!
voice-port 0/1/1
!
!
mgcp fax t38 ecm
mgcp behavior g729-variants static-pt
!
!
!
dial-peer voice 4697500 voip
translation-profile outgoing TO_CUE_VM
destination-pattern 8500
b2bua
voice-class sip outbound-proxy ipv4:192.168.0.22
session protocol sipv2
session target ipv4:192.168.0.22
session transport udp
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2 voip
translation-profile outgoing TO_CIP-TELECOM
destination-pattern [2-9]..[2-9]......$
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 3 voip
translation-profile outgoing CLID_INTL
destination-pattern 011T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:213.175.221.14
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming FROM_CIP-TELECOM
voice-class sip dtmf-relay force rtp-nte
incoming called-number 13104697570
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
!
sip-ua
authentication username 093364893 password *****
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:213.175.221.14 expires 120
sip-server dns:213.175.221.14
host-registrar
!
!
!
telephony-service
max-ephones 8
max-dn 32
ip source-address 192.168.0.38 port 2000
url services http://192.168.0.22/voiceview/common/login.do
url authentication http://192.168.0.38/CCMCIP/authenticate.asp
time-zone 5
time-format 24
voicemail 8500
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
web admin system name admin secret *****
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Jul 05 2009 13:58:28
!
!
ephone-dn 1
number 093364893
!
!
ephone-dn 3 dual-line
number 5000
description 3104697570
call-forward busy 5123640051
call-forward noan 5123640051 timeout 10
!
!
ephone-dn 4
number 5002 no-reg primary
call-forward busy 8500
call-forward noan 8500 timeout 10
!
!
ephone-dn 15
number 3598.... no-reg primary
mwi on
!
!
ephone-dn 16
number 3599.... no-reg primary
mwi off
!
!
ephone 1
device-security-mode none
mac-address 0016.46CB.4788
username \"leia\" password 12345
button 1:4
!
!
!
ephone 4
device-security-mode none
mac-address 0016.468A.E4EC
username \"perry\" password 12345
button 1:3
!
!
!
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
password *****
login local
!
scheduler allocate 20000 1000
end
Open Vpn
Open VPN client setup
Download OpenVPN 2.0.9 from
OpenVPN web site. Download prebuilt Windows installer or source distribution for Linux/Unix and Mac OS X platforms and compile it yourself.
Create file C:\Program Files\OpenVPN\config\openvpn.ovpn (/etc/openvpn/openvpn.conf on UNIX platforms) with the following data:
#--8<---------------------------------------
##############################################
# Sample client-side OpenVPN 2.0 config file #
# for connecting to multi-client server. #
# #
# This configuration can be used by multiple #
# clients, however each client should have #
# its own cert and key files. #
# #
# On Windows, you might want to rename this #
# file so it has a .ovpn extension #
##############################################
# Specify that we are a client and that we
# will be pulling certain config file directives
# from the server.
client
# Use the same setting as you are using on
# the server.
# On most systems, the VPN will not function
# unless you partially or fully disable
# the firewall for the TUN/TAP interface.
;dev tap
dev tun
# Windows needs the TAP-Win32 adapter name
# from the Network Connections panel
# if you have more than one. On XP SP2,
# you may need to disable the firewall
# for the TAP adapter.
;dev-node MyTap
# Are we connecting to a TCP or
# UDP server? Use the same setting as
# on the server.
proto tcp
;proto udp
# The hostname/IP and port of the server.
# You can have multiple remote entries
# to load balance between the servers.
remote vpn.callwithus.com 443
;remote my-server-2 1194
# Choose a random host from the remote
# list for load-balancing. Otherwise
# try hosts in the order specified.
;remote-random
# Keep trying indefinitely to resolve the
# host name of the OpenVPN server. Very useful
# on machines which are not permanently connected
# to the internet such as laptops.
resolv-retry infinite
# Most clients don't need to bind to
# a specific local port number.
nobind
# Downgrade privileges after initialization (non-Windows only)
user nobody
group nobody
# Try to preserve some state across restarts.
persist-key
persist-tun
# If you are connecting through an
# HTTP proxy to reach the actual OpenVPN
# server, put the proxy server/IP and
# port number here. See the man page
# if your proxy server requires
# authentication.
;http-proxy-retry # retry on connection failures
;http-proxy [proxy server] [proxy port #]
# Wireless networks often produce a lot
# of duplicate packets. Set this flag
# to silence duplicate packet warnings.
;mute-replay-warnings
# SSL/TLS parms.
# See the server config file for more
# description. It's best to use
# a separate .crt/.key file pair
# for each client. A single ca
# file can be used for all clients.
ca ca.crt
cert ACCOUNT.crt
key ACCOUNT.key
# Verify server certificate by checking
# that the certicate has the nsCertType
# field set to "server". This is an
# important precaution to protect against
# a potential attack discussed here:
# http://openvpn.net/howto.html#mitm
#
# To use this feature, you will need to generate
# your server certificates with the nsCertType
# field set to "server". The build-key-server
# script in the easy-rsa folder will do this.
;ns-cert-type server
# If a tls-auth key is used on the server
# then every client must also have the key.
;tls-auth ta.key 1
# Select a cryptographic cipher.
# If the cipher option is used on the server
# then you must also specify it here.
;cipher x
# Enable compression on the VPN link.
# Don't enable this unless it is also
# enabled in the server config file.
comp-lzo
# Set log file verbosity.
verb 3
# Silence repeating messages
;mute 20
#--8<---------------------------------------
In the configuration file replace ACCOUNT in key file names with your 10-digit long account number. Login into your account to request a temporary key files (valid for 2 weeks), or purchase a 1 year key. Put the files to the same directory where openvpn.ovpn file is located.
Start OpenVPN client with command "openvpn --config openvpn.ovpn". The client will print several lines on the screen, the last line should be "Initialization Sequence Completed". If you get this line, you're 99% done!
Check the connection. From another command prompt window run "ping 10.39.0.1", you should get responses from the server.
Install Xten softphone, configure it as shown on
Configuration page, but in "Domain" field enter "10.8.0.1" instead of "cip-tele.com" (without quotes). Enjoy SIP calls worldwide over blocked network:-)
The VPN will provide an access to our SIP server only and nothing else, you will not be able to access other SIP providers over our VPN link