Sip ρυθμίσεις του πελάτη

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

#--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