User Manual SIP 60X
Contents
1. 40 Page 2 of 43 SIP 60X User Manual 7 8 Figure 4 Screen Shot Of Save Configuration 41 7 9 Rebooting From Remote 41 7 10 Figure 5 Screen Shot Of Rebooting Page 41 SOFTWARE UPGRADE wast valencia hess u u S 41 RESTORE FACTORY DEFAULT SETTINGS u u 42 Page 3 of 43 SIP 60X User Manual 1 WELCOME Thank you for purchasing the SIP 60X Analog FXS IP Gateway The SIP 60X offers an easy to manage easy to configure IP communications solution for any business with virtual and or branch locations The SIP 60X supports popular voice codec s and is designed for full SIP compatibility and interoperability with 3rd party SIP providers thus enabling you to fully leverage the benefits of VoIP technology integrate a traditional phone system into a VoIP network and efficiently manage communication costs This manual will help you learn how to operate and manage your SIP 60X FXS Analog IP Gateway and make the best use of its many upgraded features including simple and quick installation multi party conferencing This IP Analog Gateway is very easy to manage and scalable specifically designed to be an easy to use and affordable VoIP solution for the small medium business or enterprise 1 1 Gateway SIP 60X Overview The SIP 60X series has a compact and qui2. 12 5 2 1 Phone or Extension 12 S229 Re eee eid c eie iu n 13 5 3 S721 tg 0 a 14 5 4 Cet secet s kal DeL t ea Dette Dt tee 14 5 5 Call Transfer ai ee ed ele ed tb ia te b te 14 5 6 Way 15 5 7 FI BUINQOGTOUD et ET 15 5 8 is as 17 5 9 PSTN Pass Thro gh Life Line rr RR 17 5 10 Sending And Receiving Fax 17 CALL a qus icc 17 CONFIGURATION GUIDBE U U u 18 7 1 Configuring SIP 60X Via Voice 18 7 2 Configuring SIP 60X With Web Browser 19 7 2 1 Access The Web Configuration Menu 19 7 3 Important uo aetate 21 7 3 1 NAT rr RED Rr e t e et Ra Rees 21 Be DTMEMethods ERO IER 22 7 3 8 Preferred VOCODER 22 7 4 End UserGoniig ratiom m ua sa ua asan a as 22 7 5 Super User Configuration 27 7 6 Figure 3 Screenshot Of Super User Configuration Login Screen 27 7 7 Saving The Configuration Changes
3. User Manual SIP 60X Analog IP Gateway 4 or 8 FXS Ports SIP 60X User Manual Table of Contents WELCOME u dese dedic Qa 4 1 1 Gateway SIP 60X OVEDIVIGW eddie dod eode eda 4 1 2 Safety COMpllances ee aaa e eue ee a ted ona 4 1 3 Warranty eee DRE RAE 5 CONFIGURE YOUR SIP GOX 2 2 Ck qawasaq cas 5 2 1 Equipment Packaging oo ORE CEDERE 5 2 2 Comnet The SIP GOK 5 2 3 Figure 1 Diagram of SIP 60X Back 6 2 4 Figure 2 Diagram Of SIP 60X Display Panel 6 APPLICATION DESCRIPTIGONN E a Rea nS Ra SR 7 3 1 Examples Of SIP 60X Configurations reete 7 3 1 1 Application SIP 60X FXS Gateway with PBX Scenario VoIP 7 3 1 2 Application SIP 60X amp SIP 60X0 Toll Free Calls Between Locations 8 SIP 60X FEATURES uuu esaet dt etude 8 4 1 Software Features Overview oto kkk ie oe e Pe EE 9 4 2 Hardware 10 BASIC OPERATIONS wiscctccenedstacdssetcentencseudscensentscesacenstensnetuescunsdenscnnedeustennteneteedtucenseste 11 5 1 Understanding Voice Prompts 11 5 2 Placing A Phone
4. G729E payload type Default value is 102 Range is from 96 to 127 VAD Default is No VAD allows detecting the absence of audio and conserve bandwidth by preventing the transmission of silent packets over the network Symmetric RTP Fax Mode Default is No When set to Yes the device will change the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device T 38 Auto Detect FoIP by default or Pass Through must use codec PGMHI PCOMA SIP 60X User Manual Fax Tone Default is Callee This decides whether Caller or Callee Detection Mode sends out the re INVITE for T 38 or Fax Pass Through Jitter Buffer Type Select either Fixed or Adaptive based on network conditions Jitter Buffer Length Select Low Medium or High based on network conditions Distinctive Ring tone Custom Ring Tone 1 to 3 with associate Caller ID when selected if Caller ID is configured then the device will ONLY uses this ring tone when the incoming call is from the Caller ID System Ring Tone is used for all other calls When selected but no Caller ID is configured the selected ring tone will be used for all incoming calls Disable Call Waiting Default is No Disable Call Waiting Tone No OYes Ring Timeout 60 10 300 seconds default is 60 seconds No Key Entry Timeout 4 seconds defaultis 4 seconds Early Dial No Yes use
5. 2 Enter 99 and wait for reset voice prompt 3 Enter 862584658050 NOTE 1 Factory Reset will be disabled if the Lock keypad update is set to Yes 2 Please be aware by default the SIP 60X WAN side HTTP access is disabled After a factory reset the device s web configuration page can be accessed only from its LAN port Page 43 of 43
6. DHCP IP Lease Time Value is set in units of hours Default value is 120 hrs 5 Days The time IP address is assigned to the LAN clients DMZ IP Forward all WAN IP traffic to a specific IP address if no matching port is used by SIP 60X or defined in port forwarding Port Map Forwards a matching TCP UDP port to a specific LAN IP address with a specific TCP UDP port End User Password This contains the password to access the Web Configuration Menu This field is case sensitive Reply to ICMP on WAN port If set to Yes the SIP 60X will respond to the PING command from other computers but it also is vulnerable to the DOS attack Default is No WAN side http access If this parameter is set to No the HTML configuration update via WAN port is disabled TABLE 8 Status Page Definitions Page 25 of 43 SIP 60X User Manual MAC Address 00 17 1 00 00 09 WAN IP Address 192 168 0 39 Product Model Unicomb5008 Software Version BOOT 1 1 0 10 2008 05 23 16 48 00 IMG 1 1 0 10 2008 05 24 14 54 00 System Up Time 0 day s 2 hour s 58 minute s 20 second s PPPoE Link Up Disabled NAT Primary Indepndent Mapping Port Dependent Filter Port Hook Registration DND Forward Busy Forward Delayed Forward FXS 1 OnHook Registered No FXS 2 OnHook Registered FXS 3 On Hook Registered No Port Status FXS 4 On Hook Registered No FX85 On Hook Registered No FXS 6 On Hook Registe
7. Syntax fl freq vol f2 freq vol c 0n1 0ff1 0n2 0ff2 0n3 0ff3 Note freq 0 4000Hz vol 30 0dBm Dial Tone f1 350 13 f2 440 13 c 0 0 Ringback Tone fi 440 19 f2 480 19 cf 2000 4 000 Call Progress Tones Reorder Tone f1 480 24 f2 620 24 c 250 250 Busy Tone f1 480 24 f2 620 24 500 500 Confirmation Tone f1 350 11 12 440 11 100 100 100 100 100 100 Call Waiting Tone 1 440 13 300 10000 300 10000 0 0 Restore Configuration Default Ring f1 440 13 f2 480 15 c 2000 4000 Cadence Only the cadence is configurable Syntax c on1 offl on2 off2 on3 off3 J i Sm Restore Configuration SUPER CONFIGURATION PAGE DEFINITIONS Setting Options Disable Voice Prompt Meaning Default is No Syslog Server The IP address or URL of System log server This feature is especially useful for the ITSP Internet Telephone Service Provider Syslog Level Select the SIP 60X to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events product model version on boot up INFO level NAT related info INFO level sent or received SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level registration status change INFO level negotiated codec INFO level Ethernet link up INFO level 9 SLIC chip exception WARNING and ERRO
8. ports 3 5 and 7 tied to SIP Account configured in Port 1 marked as Active and ports 4 6 8 tied to SIP Account configured in Port 2 marked as Active as well Example of not sequential configuration of a multiple hunt group FXS Port 1 FXS Port 2 FXS Port 3 FXS Port 4 FXS Port 5 FXS Port 6 FXS Port 7 FXS Port 8 SIP UserID and Authenticate ID entered Hunting group set to Active SIP UserlD and Authenticate ID entered Hunting Group set to Active SIP UserlD and Authenticate ID left blank Hunting Group set to 1 SIP UserlD and Authenticate ID left blank Hunting group set to 2 SIP UserlD and Authenticate ID left blank Hunting Group set to 1 SIP UserlD and Authenticate ID left blank Hunting Group set to 2 SIP UserlD and Authenticate ID left blank Hunting group set to 1 SIP UserlD and Authenticate ID left blank Hunting Group set to 2 Note A single call directed to the SIP account will NOT result in all ports ringing at the same time They will ring in the hunting group only This feature is applicable to incoming calls only Page 16 of 43 SIP 60X User Manual 5 8 Inter port Calling In some cases a user may want to make phone calls between SIP 60X gateway ports when the gateway will be used as stand alone unit without any SIP server This feature will also be applicable when the gateway is used in mode Hunting Groups and will be registered to SIP server only with one master number In
9. 3 i e 192 168 0 26 should be key in like 192168000026 No decimal is needed 5 Key entry cannot be deleted but the phone may prompt error once it is detected 5 2 Placing a Phone Call 5 2 1 Phone or Extension Numbers 1 Dial the number directly and wait for 4 seconds Default No Key Entry Timeout or Page 12 of 43 SIP 60X User Manual 2 Dial the number directly and press Use as dial key must be configured in web configuration Examples 1 Dial an extension directly on the same proxy and then press the or wait for 4 seconds 2 Dial an outside number first enter the prefix number usually 1 or international code followed by the phone number Press or wait for 4 seconds Check with your VoIP service provider for further details on prefix numbers 5 2 2 Direct IP Calls Direct IP calling allows two parties that is a FXS Port with an analog phone and another VoIP Device to talk to each other in an ad hoc fashion without a SIP proxy Elements necessary to completing a Direct IP Call 1 Both SIP 60X and other VoIP Device have public IP addresses or 2 Both SIP 60X and other VoIP Device are on the same LAN using private IP addresses or 3 Both SIP 60X and other VoIP Device can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ SIP 60X supports two ways to make Direct IP Calling Using IVR 1 Pick up the analog phone then access the voice
10. FXS8 Bana T FXS LAN WAN Power PSTN Line Reset Supply Ports TABLE 1 Definitions of SIP 60X Connectors Connect the LAN port with an Ethernet cable to your LAN PC WAN Connect to the internal LAN network or router PSTN Line 1 port Factory Reset button Press for 8 seconds to reset RESET d factory default settings DC 9V 2A Power adapter connection FXS port to be connected to analog phones fax FXS1 FXS8 i machines Once the SIP 60X is turned on and configured the front display panel indicates the status of the unit 2 4 Figure 2 Diagram of SIP 60X Display Panel Display LEDs FXS port status TABLE 2 Definitions of SIP 60X Display Panel Power LED Indicates Power Remains ON when Power is connected Page 6 of 43 SIP 60X User Manual and turned ON RUN LED Blinking after boot up LAN LED Indicates LAN port activity WAN LED Indicates WAN port activity Indicate status of the respective FXS Ports on the back panel Busy ON Solid Green Available OFF Slow blinking FXS LEDs indicates Voice Mail for that port LEDs 1 8 NOTE gt Flash blinking of RUN WAN LED together indicates a firmware upgrade or provisioning state gt LEDs POWER and WAN are ON and READY blinking when device is up and running and successfully registered to the SIP Server 3 APPLICATION DESCRIPTION There are two scenarios where
11. G711y a gt G729 A B E gt G723 gt G726 16 24 32 40 7 4 End User Configuration This section will describe the options in the Web configuration user interface As mentioned a user can log in as an administrator or end user Functions available for the end user are gt STATUS Displays the network status account status software version and MAC address of the phone BASIC OPTIONS Basic preferences such as date and time settings multi purpose keys and LCD settings can be set here Additional functions available to administrators are Super OPTIONS To set advanced network settings codec settings and XML Configuration settings PROFILE X To configure each of the SIP accounts gt FXS PORTS To configure each of the FXS ports and Hunting Groups etc TABLE 7 Basic Settings Page Definitions Page 22 of 43 SIP 60X User Manual Web 80 default for HTTP is 80 dynamically assigned via DHCP DHCP hostname optional DHCP domain optional DHCP vendor class ID optional Ouse PPPoE PPPoE account ID PPPoE password IP Address PPPoE Service Name R Preferred DNS server 0 0 10 J0 statically configured as IP Address 192 168 10 139 Subnet Mask 255 1255 14255 Default Router 192 158 10 H1 DNS Server 1 218 L2 11135 1 DNS Server 2 192 168 Ho 1 Cloned WAN MAC Addr In hex format BASIC OPTIONS SETTING Setting Options This is the device s interna
12. HTML configuration menu can be accessed via LAN or WAN port Page 19 of 43 SIP 60X User Manual From the LAN port 1 Directly connect a computer to the LAN port 2 Open a command window on the computer 3 Type in ipconfig release the IP address etc becomes 0 4 Type in ipconfig renew the computer gets an IP address 192 168 22 x segment by default 5 Open a web browser type in the default gateway IP address http 192 168 22 1 You will see the login page of the device From the WAN port The WAN port HTML configuration option is disabled by default from factory To access the HTML configuration menu from the WAN port 1 Enable the WAN Port Web Access option via IVR option 12 2 Find the WAN IP address of the SIP 60X using voice prompt menu option 02 3 Access the SIP 60X Web Configuration page by the following URI via WAN port http SIP 60X IP Address the SIP 60X IP Address is the WAN IP address for the SIP 60X NOTE If using a web browser to enter the configuration page strip the leading O s because the browser will parse in octet i e if the IP address is 192 168 001 014 please type in 192 168 1 14 Once the HTTP request is entered and sent from a Web browser the user will see a log in screen There are two default passwords for the login page User Password Web pages allowed End User Level 1234 Only Status and Basic Settings Administrator Level admin Browse all p
13. Yes only if proxy supports 484 response Dial Plan Prefix this prefix string is added to each dialed number Use as Dial Key No Yes fsetto Yes will function as the Re Dial key Dial Plan do not send SUBSCRIBE for Message Waiting Indication SUBSCRIBE for lt Yes send periodical SUBSCRIBE for Message Waiting Indication Send Anonymous 9No OYes caller ID will be blocked if setto Yes Anonymous Call Rejection No Oves Session Expiration 180 in seconds default 180 seconds Min SE 90 in seconds default and minimum 90 seconds Caller Request Timer 9No O Yes Requestfortimer when making outbound calls Callee Request Timer No O Yes When caller supports timer but did not request Force Timer No OvYes Use timer even when remote party does not support UAC Specify Refresher QUAC QOUAS Omit Recommended UAS Specify Refresher G UAC O UAS hen UAC did not specify refresher tag PROFILE PAGE DEFINITIONS Setting Options Disable Call Default is No This is to disable the stutter Call Waiting Waiting Tone Tone when a Call Waiting call arrives The CWCID will still be displayed Ring Timeout Incoming call will stop ringing when not picked up given a specific period of time No Key Entry Timeout Default is 4 seconds Early Dial Default is No Use only if proxy supports 484 response Page 36 of 43 SIP 60X User Manual This parameter cont
14. address Lock Keypad Update No Ores configuration update via keypad is disabled if setto Yes SUPER OPTIONS SETTING Setting Options This contains the password to access the Super Web Configuration page This field is case sensitive Admin Password Home NPA Local area code for North American Dial Plan This field defines the layer 3 QoS parameter which can be Layer 3 QoS the value used for IP Precedence or Diff Serv or MPLS Default value is 48 This contains the value used for layer 2 VLAN tag Default setting is blank STUN server is IP address or domain name of stun server This parameter specifies how often the SIP 60X sends a blank UDP packet to the SIP server to keep the hole on the NAT open Default is 20 seconds Default method is HTTP Firmware upgrade may take up to 10 minutes depending on network environment Do not interrupt the firmware upgrading process This parameter defines the URI or IP address of the NTP NTP Server server which is used by the SIP 60X to display the current date time If this parameter is set to Yes the configuration update via keypad is disabled Layer 2 QoS keep alive interval Firmware Upgrade and Provisioning Lock Keypad Update Page 28 of 43 Disable Voice Prompt 9 Syslog Server Syslog Level NONE SIP 60X User Manual Yes voice promptis disabled if setto Yes v Download Device Configuration
15. make calls 7 3 Important Settings The end user must configure the following settings according to the local environment NOTE Most settings on the web configuration pages are set to the default values 7 3 1 NAT Settings If you plan to keep the gateway within a private network behind a firewall we recommend using STUN Server The following three 3 settings are useful in the STUN Server scenario 1 STUN Server under Super Settings webpage Enter a STUN Server IP or FQDN that you may have or look up a free public STUN Server on the internet and enter it on this field If using Public IP keep this field blank 2 Use Random Ports under Super Settings webpage It really depends on your network settings so set this parameter to Yes or No whichever works Generally if you have multiple IP devices under the same network it should be set to Yes If using a Public IP address set this parameter to No 3 NAT Traversal under the Profile web pages Set this to Yes when gateway is behind firewall on a private network Page 21 of 43 SIP 60X User Manual 7 3 2 DTMF Methods DTMF Settings are in Profile pages gt in audio gt DTMF via RTP RFC2833 gt DTMF via SIP INFO Enable one or more DTMF methods based on your PBX system 7 3 3 Preferred VOCODER Codec The SIP 60X supports a broad range of voice codec s Under Profile web pages choose your preferred order of different codec s gt PCMU A or
16. such cases users still will be able to make inter port calls by using the IVR feature For example the user connected to port 1 can reach the user connected to port 3 by dialing and 73 Digit 7 indicated using inter port calling feature digit 3 indicates port number which should be reached At the same manner the user connected to port 4 can reach the user connected to port 8 by dialing and 78 5 9 PSTN Pass Through Life Line The RJ 11 line jack on the SIP 60X side functions as a pass through jack when the SIP 60X is out of power The pass through life line enables the user to use the analog phone for PSTN calls directly without using an access code 5 10 Sending and Receiving Fax SIP 60X supports fax in two modes 1 Fax Pass through If the service provider does not support T 38 pass through mode may be used If you have problems with sending or receiving Fax toggle the Fax Tone Detection Mode setting 2 T 38 Fax over IP 6 CALL FEATURES The SIP 60X supports the traditional telephony features available in a PBX as well as additional advanced telephony features TABLE 6 Call Features Table Star Code Key Call Features 30 Block Caller ID for all config change 31 Send Caller ID for all config change 67 Block Caller ID per call 82 Send Caller ID per call 47 Direct IP Calling Dial 47 IP address No dial tone will be Page 17 of 43 SIP 60X User Manual played i
17. switch to a new channel for a new call Pressing pound sign will serve as Re Dial key 7 CONFIGURATION GUIDE 7 1 Configuring SIP 60X via Voice Prompt DHCP MODE Select voice menu option 01 to enable SIP 60X to use DHCP Page 18 of 43 SIP 60X User Manual STATIC IP MODE Select voice menu option 01 to enable SIP 60X to use STATIC IP mode then use option 02 03 04 05 to set up IP address Subnet Mask Gateway and DNS server respectively FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server UPGRADE PROTOCOL Select voice menu option 15 to choose firmware and configuration upgrade protocol User can choose between TFTP and HTTP FIRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode among the following three options 1 Always check 2 Check when pre suffix changes and 3 Never upgrade WAN PORT WEB ACCESS Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages 7 2 Configuring SIP 60X with Web Browser SIP 60X has an embedded Web server that will resoond to HTTP GET POST requests It also has embedded HTML pages that allow users to configure the SIP 60X through a Web browser such as Microsoft s IE and AOL s Netscape 7 2 1 Access the Web Configuration Menu The SIP 60X
18. the SIP 60X series can be effectively used to enable any business to leverage the benefits of VoIP and the Internet 3 1 Examples of SIP 60X Configurations 3 1 1 Application SIP 60X FXS Gateway with PBX Scenario Anywhere in the world FXS Trunks Analog Phones 4 8 Ports INTERNET CLOUD Page 7 of 43 SIP 60X User Manual 3 1 2 Application SIP 60X amp SIP 60X0 Scenario Toll Free Calls between Locations ITISP1 ITISP2 udi 00000000 0 e 4 Internet Branch B 4 SIP 60X FEATURES The SIP60 x is a next generation IP voice gateway that is interoperable and compatible with leading IP PBXs SoftSwitches and SIP platforms The SIP 60X FXS series is auto configurable remotely manageable and scalable There are two FXS models the SIP 604 and SIP 608 each offering superb voice quality traditional telephony functionality easy deployment and 4 or 8 FXS ports respectively Each model features flexible dialing plans PSTN failover integrated call routing to support a pure IP network call and an external power supply Page 8 of 43 SIP 60X User Manual 4 1 Software Features Overview 4 or 8 FXS ports Two RJ 45 ports switched or routed Multiple SIP accounts amp profiles 4 or 8 accounts choice of 2 profiles per account Supports Voice Codec s G711 a u Annex amp 11 G723 1A G726 ADPCM with 16 24 32 40 bit rates G729 A B E fax pass throu
19. 0 non condensing Compliance FCC CE Page 10 of 43 SIP 60X User Manual 5 BASIC OPERATIONS 5 1 Understanding SIP 60X Voice Prompts SIP 60X has a stored voice prompt menu for quick browsing and simple configuration To enter the voice prompt menu press FXS port kkk on the standard analog phone connected to any TABLE 5 Definitions of SIP 60X Voice Prompts Menu Voice Will Say the Following Main Enter a Menu Option Enter for the next menu option Menu Enter to return to the main menu Enter 01 05 07 10 17 47 86 or 99 Menu option 01 DHCP Mode PPPoE Enter 9 to toggle the selection Mode or Static IP Mode If user selects Static IP Mode user need configure all the IP address information through menu 02 to 05 If user selects Dynamic IP Mode the device will retrieve all IP address information from DHCP server automatically when user reboots the device 02 IP Address IP address The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode 03 Subnet IP address Same as Menu option 02 04 Gateway IP address Same as Menu option 02 05 DNS Server IP address Same as Menu option 02 07 Preferred Vocoder Enter 9 to go to the next selection in the list gt PCMU gt PCMA gt iLBC gt G 726 gt G 723 gt G 729 10 MAC Address Announces the Mac address o
20. Allows the user to configure a User ID or extension number to be automatically dialed upon off hook Only the user part of a SIP address needs to be entered here The phone will automatically append the and the host portion of the corresponding SIP address Proxy Require SIP Extension to notify SIP server that the unit is behind the NAT Firewall Use NAT IP NAT IP address used in SIP SDP message Default is blank Disable Call Waiting Default is No No Key Entry Timeout Default is 4 seconds Preferred Vocoder The SIP 60X supports up to 5 different Vocoder types including G 711 A U law G 726 Supports bit rates 16 24 Page 33 of 43 SIP 60X User Manual 32 and 40 G 723 1 G 729A B E and iLBC The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message The first Vocoder is entered by choosing the appropriate option in Choice 1 The last Vocoder is entered by choosing the appropriate option in Choice 8 Ring Tone 1 RingTone Ring Tone1 N v Page 34 of 43 SIP 60X User Manual PROFILE PAGE DEFINITIONS Setting Options Voice Frames per TX Meaning This field contains the number of voice frames to be transmitted in a single packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configu
21. F Finland Sweden DTMF Denmark On hook Voltage Select the on hook voltage to suit different area or PBX Polarity Reversal Select Polarity Reversal to adapt some call charge billing system Default is No Hook Flash Timing Time period when the cradle is pressed Hook Flash to simulate FLASH To prevent unwanted activation of the Flash Hold and automatic phone ring back adjust this time value Volume Amplification Handset volume adjustment RX is for receiving volume TX is for transmission volume Default values are OdB for both parameters 6dB generates the highest volume and 6dB generates the lowest volume Ring Tones This function lets you configure ring tone cadence preferences User has 10 choices The configuration completed in Distinctive Ring Tones block in the same page applies to ring tones cadences configured here 7 7 Saving the Configuration Changes Once a change is made press the Update button in the Configuration Menu The following screen will confirm that the changes have been saved To activate changes reboot or power cycle the SIP 60X after changes are made Page 40 of 43 SIP 60X User Manual 7 8 Figure 4 Screen Shot Of Save Configuration Page Your changes have been saved Please wait 5 second and then reboot the device Configuration DEVICE STATUS BASIC OPTIONS SUPER OPTIONS PROFILE 1 PROFILE 2 FXS PORTS 7 9 Rebooti
22. R levels 10 memory exception ERROR level ON Oo m NS Download Device Configuration User can download configuration from the web page and save to configuration file Call Progress Tones Using these settings user can configure tone frequencies Page 29 of 43 SIP 60X User Manual according to their preference By default they are set to North American frequencies Frequencies should be configured with known values to avoid uncomfortable high pitch sounds ON is the period of ringing On time in ms while OFF is the period of silence In order to set a continuous tone OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Restore Configuration User can restore the before configuration from the configuration file saved at local pc TABLE 10 FXS Ports Configuration Definitions FXS Port SIP User ID 1 86033 2 86019 3 8205 4 8206 5 8207 8 8208 7 8209 8 8210 Setting Options FXS Port Authenticate ID Password Name Profile ID 85033 Profile 1 85019 Profile 1 8205 Profile 1 8206 Profile 1 8207 Profile 1 8208 Profile 1 8209 Profile 1 8210 Profile 1 SaveSet Reboot FXS PORT SETTING Meaning FXS Port Number SIP User ID User account information provided by VoIP service provider ITSP Usually in the form of digit similar to phone number or actually a phone number Au
23. T is detected STUN will not work and ONLY outbound proxy can correct the problem NAT Traversal This parameter defines whether the SIP 60X NAT traversal mechanism is activated or not If activated by choosing Yes and a STUN server is also specified then the SIP 60X performs according to the STUN client specification Under this mode the embedded STUN client will detect if and what type of firewall NAT is being used If the detected NAT is a Full Cone Restricted Cone ora Port Restricted Cone the SIP 60X will use its mapped public IP address and port in all of its SIP and SDP messages If the NAT Traversal field is set to Yes with no specified STUN server the SIP 60X will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open Page 31 of 43 SIP 60X User Manual If your home or office router can act as a UPNP server you can select UPNP option for NAT traversal Use DNS SRV Default is No If set to Yes the client will use DNS SRV to look up server User ID is Phone Number If the SIP 60X has an assigned PSTN telephone number this field should be set to Yes Otherwise set it to If Yes is set a parameter will be attached to the From header in SIP request SIP Registration This parameter controls whether the SIP 60X needs to send REGISTER
24. ages Only an administrator can access the SUPER SETTINGS configuration page 1 There are six different tabs STATUS Basic Settings SUPER Settings Profile 1 Profile 2 and FXS Ports on the top of the screen after login To open each page click on the tab 2 Click on Profile 1 to enter your SIP Server SIP Proxy Registrar information Enter the IP Address or FQDN of the Server under SIP Server and or Outbound Proxy 3 Click on FXS ports to enter the extensions or account information You will need to fill in the following information for each extension Once the extensions are configured you Poge 20 of 43 SIP 60X User Manual are finished FXS PORT SEEGERS 210 210 210 Port SIP User ID Authenticate ID Password Name Profile ID Hunting Group Profile 1 Active 211 211 211 Profile 1 212 212 212 Profile 1 213 213 213 Profile 1 Active 214 214 214 Profile 1 215 215 215 Profile 1 iim 216 216 216 Profile 1 EN lt 21T 21T 21T Profile 1 4 Click save set after changing any setting and then Re boot to confirm changes 5 After reboot check the Status Page to confirm the extensions are successfully registered You can now use your standard phones connected to ports FXS1 to FXS8 to
25. d on a LAN Users are recommended to conduct TFTP upgrade in a controlled LAN environment if possible NOTES When SIP 60X boots up it will send TFTP or HTTP request to download configuration files there are two configuration files one is cfg bin and the other is cfg001fc1xxxxxx where 001fc1xxxxxx is the MAC address of the SIP 60X These two files are for initial automatically provisioning purpose only for normal TFTP or HTTP firmware upgrade the following error messages in a TFTP or HTTP server log can be ignored 9 RESTORE FACTORY DEFAULT SETTINGS WARNING Restoring the Factory Default Setting will DELETE all configuration information of the phone Please BACKUP or PRINT out all the settings before you approach to following steps Netronix will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider FACTORY RESET There are two 2 methods for resetting your unit Reset Button Reset default factory settings following these four 4 steps 1 Unplug the Ethernet cable 2 Locate a needle sized hole on the back panel of the gateway unit next to the power connection 3 Insert a pin in this hole and press for about 8 seconds 4 Take out the pin All unit settings are restored to factory settings Page 42 of 43 SIP 60X User Manual IVR Command Reset default factory settings using the IVR Prompt Table 5 1 Dial for voice prompt
26. et design no fans and offers superb audio quality rich feature functionality strong security protection and good manageability It is auto configurable remotely manageable and scalable The SIP 60X features 4 or 8 port FXS interface for analog telephones dual 10M 100Mbps network ports with integrated router PSTN life line in case of power failure In addition it supports the option of 2 SIP Server profiles caller for various countries regions T 38 fax flexible dialing plans security protection SIPS TLS comprehensive voice codec s including G 711 a u law G 723 1 G 726 16 24 32 48 bit rates G 729A B E Caution Changes or modifications to this product not expressly approved by the manufacturer or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Information in this document is subject to change without notice No part of this document may be reproduced or transmitted in any form or by any means electronic or mechanical for any purpose without the express written permission of the manufacturer 1 2 Safety Compliances The SIP 60X is compliant with various safety standards including FCC CE Its power adaptor is compliant with UL standard Warning use only the power adapter included in the SIP 60X package Using an alternative power adapter may permanently damage the unit Page 4 of 43 SIP 60X User Manual 1 3 Warranty Netronix markets its p
27. f the unit 12 WAN Port Web Access Enter 9 to toggle between enable and disable Page 11 of 43 SIP 60X User Manual 13 Firmware Server IP Announces current Firmware Server IP Address address Enter 12 digit new IP address 14 Configuration Server IP Announces current Config Server Path Address IP address Enter 12 digit new IP address 15 Upgrade Protocol Upgrade protocol for firmware and configuration update Enter 9 to toggle between TFTP and HTTP 16 Firmware Version Firmware version information 17 Firmware Upgrade Firmware upgrade mode Enter 9 to rotate among the following three options 1 always check 2 check when pre suffix changes 3 never upgrade 47 Direct IP Calling Enter the target IP address to make a direct IP call after dialtone See Make a Direct IP Call 99 RESET Enter 9 to reboot the device or Enter MAC address to restore factory default setting See Restore Factory Default Setting section Invalid Entry Automatically returns to Main Menu Five Success Tips when using the Voice Prompt 1 shifts down to the next menu option 2 returns to the main menu 3 9 functions as the ENTER key in many cases to confirm an option 4 All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address For IP address add 0 before the digits if the digits are less than
28. gh and T 38 Fax Comprehensive Dial Plan support for Outgoing calls G 168 Echo Cancellation Voice Activation Detection VAD Comfort Noise Generation CNG and Packet Loss Concealment PLC gt Supports PSTN PBX analog telephone sets or analog trunks Y V ON v v v TABLE 3 SIP 60X SOFTWARE FEATURES SIP FXS Analog Gateway Series SIP 604 4 ports 4 SIP accounts amp choice of 2 profiles Telephone Interfaces SIP 608 8 ports 8 SIP accounts amp choice of 2 profiles FXS RJ 11 Network Interface Two 2 10M 100 Mbps RJ 45 LED Indicators Power and Line LEDs Voice Activity Detection VAD with CNG comfort noise generation and PLC packet loss concealment AEC Voice over Packet with NLP Packetized Voice Protocol Unit supports Capabilities RTP RTCP and AAL2 protocol G 168 compliant Echo Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 PSTN Fail over PSTN failover port on power failure G 711 Annex PLC Annex VAD CNG format encoder and decoder G 723 1A G 726 ADPCM with 16 24 32 40 bit rates G 729A B E iLBC G 726 provides proprietary VAD CNG and signal power estimation Voice Play Out unit reordering fixed and adaptive jitter buffer clock synchronization AGC automatic gain control Status output Decoder controlling via voice packet header Voice Compression DHCP Server Client Yes NAT Router or Switched Mode T 38 c
29. ing digits 1617 Example 2 1900x lt 1617 gt xxxxxxx Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers Example 3 1 2 9 lt 2 011 gt x Allow any length of number with leading digit 2 and 10 digit numbers of leading digit 1 and leading exchange number between 2 and 9 if leading digit is 2 replace leading digit 2 with 01 1 before dialing 3 Default Outgoing x Example of a simple dial plan used in a Home Office in the US 1900 lt 1617 gt 2 9 1 2 9 2 9 011 2 9 x 3469 1 1 o Page 37 of 43 SIP 60X User Manual Explanation of example rule reading from left to right 1900x prevents dialing any number started with 1900 lt 1617 gt 2 9 allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length 011 2 9 x allows international calls starting with 011 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Note In some cases user wishes to dial strings such as 123 to activate voice mail or other application provided by service provider In this case should be predefined inside dial plan feature and the Dial Plan should be x Subscribe for MWI Default is No When set to Yes a SUBSCRIBE f
30. l HTTP server port Default is 80 There are two modes to operate the SIP 60X DHCP mode all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The SIP 60X acquires its IP address from the first DHCP server it discovers from the LAN it is connected IP Address Using the PPPoE feature set the PPPoE account settings The SIP 60X will establish a PPPoE session if any of the PPPoE fields is set Web Port Static IP mode configure the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary fields These fields are set to zero by default Cloned WAN MAC Allow the user to set a specific MAC address Addr Set in Hex format Page 23 of 43 SIP 60X User Manual Time Zone GMT 8 00 Beijing Singapore Taipei Kuala Lumpur Irkutsk Perth No Daylight Savings om O Yes if setto Yes display time will be 1 hour ahead of normal time Year Month Day Date Display Format Manth Day Year Day Manth Year BASIC OPTIONS SETTING Setting Options 2 Controls how the date time is displayed according to the specified time zone This parameter controls whether the displayed time will Daylight Savings Time be daylight savings time or not If set to Yes then the displayed time will be 1 hour ahead of normal time Allow user to choose among the following three format
31. lable use hook flash toggle on off hook quickly You may drop a call using hook flash 5 4 Call Waiting Call waiting tone 3 short beeps indicates an incoming call if the call waiting feature is enabled Toggle between incoming call and current call by pressing the flash button First call is placed on hold Press the flash button to toggle between two active calls 5 5 Call Transfer Blind Transfer Assume that call Caller A and B are in conversation A wants to Blind Transfer B to C 3 Caller A presses FLASH on the analog phone to hear the dial tone 4 Caller A dials 87 then dials caller C s number and then or wait for 4 seconds 5 Caller A will hear the confirm tone Then A can hang up NOTE Enable Call Feature must be set to Yes in web configuration page Caller A can place a call on hold and wait for one of three situations 1 A quick confirmation tone similar to call waiting tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point Caller A can either hang up or make another call 2 A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response for the INVITE and we will try to recover the call The busy tone is just to indicate to the transferor that the transfer has failed 3 Continuous busy tone The phone has timed out Note c
32. menu prompt by dial 2 Dial 47 to access the direct IP call menu 3 Enter the IP address using format ex 192 168 0 160 after the dial tone Using Star Code 1 Pick up the analog phone then dial 47 2 Enter the target IP address using same format as above Note NO dial tone will be played between step 1 and 2 Destination ports can be specified by using number 5 encoding for followed by the port Examples a If the target IP address is 192 168 0 160 the dialing convention is 47 or Voice Prompt with option 47 then 192 168 0 160 Followed by pressing the key if it is configured as a send key or wait 4 seconds In this case the default destination port 5060 is used if no port is specified b If the target IP address port is 192 168 1 20 5062 then the dialing convention would be 47 or Voice Prompt with option 47 then 192 168 0 160 5062 followed by pressing the key if it is configured as a send key or wait for 4 seconds NOTE When completing direct IP call the Use Random Port should set to NO You cannot make direct IP calls between FXS1 to FXS2 since they are using same IP Page 13 of 43 SIP 60X User Manual 5 3 Call Hold Place a call on hold by pressing the flash button on the analog phone if the phone has that button Press the flash button again to release the previously held Caller and resume conversation If no flash button is avai
33. messages to the proxy server The default setting is Yes Unregister on Default is No If set to Yes the SIP user s registration Reboot information is cleared on reboot Register Allows the user to specify the time frequency in minutes Expiration for the SIP 60X to refresh its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days Outgoing Call Default No If set to Yes user can place outgoing without Registration calls even when not registered if allowed by ITSP but is unable to receive incoming calls Local SIP port Defines the local SIP port the SIP 60X will listen and transmit The default value for Profile 1 5060 and 6060 for Profile 2 Local RTP Port Defines the local RTP RTCP port pair the PROFILE will listen and transmit It is the base RTP port for channel 0 When configured channel 0 will use this value for RTP and the port value 1 for its RTCP channel 1 will use port_value 2 for RTP port_value 3 for its RTCP and so on The default value for Profile 1 is 5004 and 6004 for Profile 2 Use random port This parameter when set to YES will force random generation of both the local SIP and RTP ports This is usually necessary when multiple SIP 60X are behind the same NAT Refer to Use Target Contact Default is No If set to Yes then for Attended Tran
34. n the middle Detail see Direct IP Calling section on page 12 50 Disable Call Waiting for all config change 51 Enable Call Waiting for all config change 69 Call Return Service Dial 69 and the phone will dial the last incoming phone number received 70 Disable Call Waiting Per Call 71 Enable Call Waiting Per Call 72 Unconditional Call Forward Dial 72 and then the forwarding number followed by Wait for dial tone and hang up dial tone indicates successful forward 73 Cancel Unconditional Call Forward Dial 73 and wait for dial tone then hang up 74 Enable Paging Call Dial 74 and then the destination phone number you want to activate in Paging mode 78 Enable Do Not Disturb DND When enabled all incoming calls will be rejected 79 Disable Do Not Disturb DND When disabled incoming calls will be accepted 87 Blind Transfer 90 Busy Call Forward Dial 90 and then the forwarding number followed by Wait for dial tone then hang up 91 Cancel Busy Call Forward Dial 91 Wait for dial tone Hang up 92 Delayed Call Forward Dial 92 and then the forwarding number followed by Wait for dial tone then hang up 93 Cancel Delayed Call Forward Dial 93 for a dial tone then hang up Flash Hook f user hears call waiting beep flash hook will switch to the new incoming call Also used to
35. nd after A hangs up 5 7 Hunting Group This feature allows user to setup a single SIP account on the gateway and have the ability to use all FXS ports to make receive calls Using this feature all ports active in same hunt group will have the same phone number and incoming calls will be distributed in a round robin manner among the ports active in that hunt group The number of hunting groups is limited by the number of ports each SIP 60X gateway model has i e each port can be its own hunt group The most practical and efficient way to use hunt groups is to assign 2 or 3 ports to separate hunt groups One additional and popular way to use the Hunting Group feature is called multiplexed analog lines In this configuration a legacy PBX system with 8 FXO trunks can be connected to 8 SIP 608 ports configured as a hunt group The SIP 608 can be registered to a SIP server provider using only one phone number If the SIP service provider allows multiple calls to the same number the SIP 608 will allow 8 concurrent calls to the same SIP number All office members can be reached remotely using the same phone number Page 15 of 43 SIP 60X User Manual in round robin fashion Example Configuration of a typical Hunting Group 1 Configure the SIP account from your VoIP Service Provider on FXS port 1 under FXS Ports webpage 2 Select Active under the Hunting Group drop box for FXS port 1 3 For the remaining ports say 2 3 and 4 select 1 for H
36. ng From Remote The administrator can remotely reboot the unit by pressing the Reboot button at the bottom of the configuration menu The user can re login to the unit after waiting for about 30 seconds 7 10 Figure 5 Screen Shot of Rebooting Page Click to relogin 8 SOFTWARE UPGRADE To upgrade software SIP 60X can be configured with a TFTP server where the new code image is located The TFTP upgrade can work in either static IP or DHCP mode using private or public IP address It is recommended to set the TFTP server address in either a public IP address or on the same LAN with the SIP 60X There are two ways to set up the TFTP server to upgrade the firmware namely through voice menu prompt or via the SIP 60X s Web configuration interface To configure the TFTP server via voice prompt follow section 5 1 with option 06 once set up the TFTP IP Page 41 of 43 SIP 60X User Manual address power cycle the ATA the firmware will be fetched once the ATA boots up To configure the TFTP server via the Web configuration interface open up your browser to point at the IP address of the SIP 60X Input the admin password to enter the configuration screen From there enter the TFTP server address in the designated field towards the bottom of the configuration screen Once the TFTP server is configured please power cycle the SIP 60X TFTP process may take as long as 1 to 2 minutes over the Internet or just 30 seconds if it is performe
37. ompliant Group 3 Fax Relay up to 14 4kpbs and Fax over IP auto switch to G 711 for Fax Pass through Fax Datapump V 17 V 19 V 27ter V 29 for T 38 fax relay QoS Diffserve TOS 802 1 P Q VLAN tagging IP Transport RTP RTCP Page 9 of 43 SIP 60X User Manual flexible DTMF transmission method User interface of pIMr Merhod In audio RFC2833 and or SIP Info IP Signaling SIP RFC 3261 Provisioning TFTP HTTP HTTPS pending Control TLS SIPS Management Syslog support BES pending Telnet remote management using Web browser Dial Plan Yes UPnP Support Yes Power Output 9VDC Input 100 240 VAC 50 60 Hz Mounting Rack mount Wall mount Desktop Short and long haul RENS Up to150 fton 24 AWG line Caller ID Bellcore Type 1 amp 2 ETSI BT NTT and DTMF based CID Polarity Reversal Wink Yes EMC EN55022 EN55024 and FCC part15 Class B Safety UL 4 2 Hardware specification The hardware specifications of the SIP FXS series are detailed in Table 4 TABLE 4 Hardware Specification of SIP 60X Ports 4 or 8 FXS Ports LAN interface 2 x RJ45 10 100Mbps switched or routed PSTN Port PSTN fail over port LED 4or8LEDs GREEN Universal Switching Input 100 240V AC 50 60Hz 0 5A Max Power Adaptor Output 9V DC 2A UL certified Dimension 225mm L x 135mm W x 35mm H Weight 0 29 Ibs 3 5 oz Temperature 32 104 F 0 40 C Humidity 10 9
38. ontinuous busy tone does not indicate the transfer has been successful nor does it indicate the transfer has failed It often means there was a failure to receive second NOTIFY check firmware for most recent release Attended Transfer Assume that Caller A and B are in conversation Caller A wants to Attend Transfer B to C 1 Caller A presses FLASH on the analog phone for dial tone Page 14 of 43 SIP 60X User Manual 2 Caller A then dials Caller C s number followed by or wait for 4 seconds 3 If Caller C answers the call Caller A and Caller C are in conversation Then A can hang up to complete transfer 4 Caller does not answer the call Caller A can press flash to resume call with Caller B NOTE When Attended Transfer fails and A hangs up the SIP 60X will ring back user A to remind A that B is still on the call A can pick up the phone to resume conversation with B 5 6 3 Way Conferencing The SIP 60X supports Bellcore style 3 way Conference Instructions for 3 way conference Assuming that call party A and B are in conversation A SIP 60X wants to bring C ina conference 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 A dials 23 C s number then or wait for 4 seconds 3 If C answers the call then A presses FLASH to bring B C in the conference 4 If C does not answer the call can press FLASH back to talk to B 5 Conference e
39. or Message Waiting Indication will be sent periodically Send Anonymous If this parameter is set to Yes the From header along with Privacy and Asserted Identity headers in outgoing INVITE message will be set to anonymous blocking Caller ID Anonymous Call Default is No If set to Yes incoming calls with Rejection anonymous Caller ID will be rejected with 600X Busy message zn Default is 180 seconds Expiration Min SE Default is 90 seconds Caller Request Default is NO Timer Callee Request Timer Default is NO Force Timer Default is NO Specify Default is Omit Refresher JAS Specify Default is UAC Refresher Page 38 of 43 SIP 60X User Manual 9 Standard 600 Ohm North America Bellcore North America ewo 098 default v Rx 0dB default x t5 m Page 39 of 43 Setting Options Force INVITE SIP 60X User Manual PROFILE PAGE DEFINITIONS Default is NO Special Feature Default is Standard Choose the selection to meet some special requirements from Soft Switch vendors like Nortel Broadsoft etc FXS Impedance Selects the impedance of the analog telephone connected to the Phone port Caller ID Scheme Select the Caller ID Scheme to suit the standard of different area Bellcore North America ETSI FSK France Germany Norway Taiwan UK CCA ETSI DTM
40. red No FXS 7 On Hook Registered No FXS 8 On Hook Registered No EN tt i ia s STATUS PAGE DEFINITIONS Setting Options Meaning The device ID in HEX format This is needed for ISP troubleshooting Note there are separate MAC addresses for the WAN side and the LAN side MAC Address WAN IP Address Shows WAN IP address of SIP 60X Product Model Contains the product model info Software Version Program This is the main software release Boot and Loader are not changed often System Up Time Shows system up time since the last reboot PPPoE Link Up Shows whether the PPPoE connection is running if connected to DSL modem NAT Shows type of NAT the SIP 60X is connected to via its WAN port Itis based on STUN protocol Port Status Shows several information regarding the individual FXS ports Ex Port Hook Registration DND Forward Busy Delayed Forward Forward FXS1 On Hook Registered No 613 FXS2 Off Hook Registered No 614 FXS3 On Hook Not Registered No FX 4 On Hook Registered Yes 615 EXS port 4 user has set Do Not Disturb Page 26 of 43 SIP 60X User Manual FXS port 1 user has set his calls to be forwarded unconditionally to ext 613 FXS port 2 users have set his calls to be forwarded to 614 when his phone is busy FXS port 3 users are not registered with his SIP Server Super User configuration include
41. ring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first vocoder is configured as G723 and the Voice Frames is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the SIP 60X will use and save the maximum allowed value for the corresponding first vocoder choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively G723 Rate Defines the encoding rate for G 723 vocoder By default 6 3kbps rate is chosen iLBC Frame Size Sets the iLBC frame size in 20ms or 30ms iLBC Payload type Defines payload type for iLBC Default value is 97 The valid range is between 96 and 127 G726 16 Payload type Default value is 98 Range is from 96 to 127 G726 24 Payload type Default value is 99 Range is from 96 to 127 G726 40 Payload type Default value is 103 Range is from 96 to 127
42. roducts through reseller partner channels end users should contact the company from whom you purchased the product for replacement or repair If you purchased the product directly from Netronix contact your Netronix Sales and Service Representative for a RMA Return Materials Authorization number Netronix reserves the right to remedy warranty policy without prior notification 2 CONFIGURE YOUR SIP 60X Connecting your SIP 60X is easy Before you begin please verify the contents of the SIP 60X package 2 1 Equipment Packaging Unpack and check all accessories The SIP 60X package contains gt SIP 60X VoIP adapter gt universal power supply gt One Ethernet cable 2 2 Connect the SIP 60X Managing the SIP 60X gateway and connecting the unit to the VoIP network is very simple Follow these four 4 steps to connect your SIP 60X gateway to the Internet and access the unit s configuration pages 1 Connect standard touch tone analog phones to the FXS1 FXS8 ports 2 Insert the Ethernet cable into the WAN port of SIP 60X and connect the other end of the Ethernet cable to an uplink port a router or a modem etc 3 Connect a PC to the LAN of SIP 60X for initial configuration or if it is being used as a router 4 Plug the power adapter into the SIP 60X and into a power outlet Page 5 of 43 SIP 60X User Manual 2 3 Figure 1 Diagram of SIP 60X Back Panel LINE FXS FXS2 FXS3 FXS4 FXS FXS
43. rols whether the phone will send an early INVITE each time a key is pressed when a user dials a number If set to Yes an INVITE is sent using the dial number collected thus far Otherwise no INVITE is sent until the Re Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re Dial button The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address responses Otherwise the call will likely be rejected by the proxy with a 404 Not Found error This feature is NOT designed to work with and should NOT be enabled for direct IP to IP calling Dial Plan Prefix Sets the prefix added to each dialed number Use as Dial Key Allows users to configure the key as the Send or Dial key If set to Yes will send the number In this case this key is essentially equivalent to the Dial key this key can be included as part of number Dial Plan Dial Plan Rules 1 Accept Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 at least 2 digits number Xx at least 2 digits number exclude 8 5 any digit of 3 4 or 5 147 any digit 1 4 or 7 f 22011 replace digit 2 with 011 when dialing Example 1 369 11 1617xxxxxxx Allow 311 611 911 and any 10 digit numbers of lead
44. s Year Month Day Month Day Year Day Month Year Date Display Format Device Mode QNATRouter O Bridge LAN Subnet Mask 255 255 2550 Default is 255 255 255 0 LAN DHCP Base IP 192 168 2 1 Base IP for the LAN port default 192 168 2 1 DHCP IP Lease Time 24 Hours Default is 120 hours or 5 days DMZ IP WAN LAN IP LAN Protocol UDP WAN LAN IP LAN Port Protocol UDP lt WAN Port LANIP LAN Port Protocol UDP v WAN Port LAN IP LAN Port Protocol UDP WAN LAN IP LAN Protocol UDP WAN LAN IP LAN Port Protocol UDP gt WAN Port LAN IP LAN Port Protocol UDP WAN LAN IP LAN Protocol UDP End User Password Basic user password to configure this device Reply to ICMP on WAN port No Yes Unitwill not respond to PING from WAN side if setto No WAN side http access O Yes WAN side access to http server will be rejected if set to No jw Poge 24 of 43 Setting Options Device Mode SIP 60X User Manual BASIC OPTIONS SETTING Meaning This parameter controls whether the device is working in NAT router mode or Bridge mode Save the setting and reboot prior to configuring the SIP60X LAN Subnet Mask Sets the LAN subnet mask Default value is 255 255 255 0 LAN DHCP Base IP Base IP for the LAN port which functions as a Gateway for the subnet Default value is 192 168 22 1
45. s not only the end user configuration but also super configurations such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration 7 5 Super User Configuration Log in to the Super User Configuration Page the same way as for the basic configuration page Log in using either of the following passwords admin or 123 7 6 Figure 3 Screenshot of Super User Configuration Login Screen 771 Super User configuration includes the end user configuration and Super configurations including SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration TABLE 9 Super Configuration Page Definitions Page 27 of 43 SIP 60X User Manual Admin Password purposely not displayed for security protection Home NPA Layer 3 905 48 Diff Berv or Precedence value Layer 2 QoS 802 1 QVLAN Tag 802 1p priority value 0 7 STUN server is R URI or IP port keep alive interval 20 seconds default 20 seconds Upgrade Via TFTP HTTP Firmware Server Path 192 168 0 159 Config Sewer Path 192 168 0 169 Firmware Upgrade Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix and Provisioning Automatic Upgrade No Yes check for upgrade every 7 minutes default 7 days Always Check for New Firmware Check New Firmware only when FAN pre suffix changes NTP Server time gist gov URI or IP
46. sfer the Refer To header uses the transferred target s Contact header information Page 32 of 43 DTMF Payload Type 1101 DTMF in Audio ONo DTMF via RFC2833 ONo DTMF via SIP INFO 9 Send Flash Event Enable Call Features ONo Offhook Auto Dial Proxy Require Use NAT IP Disable Call Waiting O No Key Entry Timeout 4 choice 1 choice 2 Preferred Vocoder choice 3 in listed order choice 4 choice 5 choice 6 SIP 60X User Manual 9 Yes Yes Oves Yes Flash will be sent as eventif setto Yes 9 Yes if Yes call features using star codes will be supported locally User ID extension to dial automatically when offhook used in SIP SDP message if specified O Yes in seconds default is 4 seconds current setting is PCMU current setting is G 726 32 v current setting is 723 1 v be current setting is G 728 current setting is G 729A B v current setting is PROFILE PAGE DEFINITIONS Setting Options DTMF Payload Type Meaning Sets the payload type for DTMF using RFC2833 DTMF in audio Send DTMF as inband in audio DTMF via RFC2833 Default YES DTMF via SIP INFO Send DTMF via SIP INFO message Send Flash Default is No If set to yes flash will be sent as DTMF Event event Enable Call Default is Yes If Yes call features using star codes will Features be supported locally Off Hook Auto Dial
47. thenticate ID SIP service subscribers Authenticate ID used for authentication Can be identical to or different from SIP User ID Password SIP service subscribers account password for SIP 60X to register to SIP servers of ITSP Name Name Profile ID Select the corresponding Profile ID 1 2 TABLE 11 Profile Page Definitions Page 30 of 43 Account Active SIP Server Outbound Proxy NAT Traversal Use DNS SRV User ID is phone number SIP Registration Unregister On Reboot Register Expiration Outgoing Call without Registration local SIP port local RTP port Use random port Refer To Use Target Contact SIP 60X User Manual Yves e 9 sip mycompany com or IP address 192 168 0 7 e g proxy myprovider com or IP address if any No No butsendkeep alive OSTUN OUPNP G No No O Yes ONo Yes 9No OYes 2 tin minutes default 1 hour max 45 days 9No 5060 default 5060 5004 1024 65535 default 5004 Oves No Oves PROFILE PAGE DEFINITIONS Setting Options Account Active Meaning When set to Yes the SIP Profile is activated SIP Server SIP Server s IP address or Domain name provided by VoIP service provider Outbound Proxy IP address or Domain name of Outbound Proxy or Media Gateway or Session Border Controller Used by SIP 60X for firewall or NAT penetration in different network environments If symmetric NA
48. unting Group Ports 2 3 and 4 are now active members of the hunting group associated with port 1 This configuration will route all calls directed to FXS port 1 to ports 2 3 and or 4 in round robin fashion respectively if port 1 is busy You can configure the ring timeout on the Profile page Example configuration of a multiple hunt group FXS Port 1 FXS Port 2 FXS Port 3 FXS Port 4 FXS Port 5 FXS Port 6 FXS Port 7 FXS Port 8 SIP UserID and Authenticate ID entered Hunting group set to Active SIP UserID and Authenticate ID left blank Hunting Group set to 1 SIP UserID and Authenticate ID left blank Hunting Group set to 1 SIP UserID and Authenticate ID entered Hunting group set to Active SIP UserID and Authenticate ID left blank Hunting Group set to 4 SIP UserID and Authenticate ID left blank Hunting Group set to 4 SIP UserID and Authenticate ID entered Hunting group set to Active SIP UserID and Authenticate ID left blank Hunting Group set to 7 Hunt Group 1 contains ports 1 2 3 Hunt Group 4 contains ports 4 5 6 Hunt Group 7 contains ports 7 8 Please be aware the choice of 1 for ports 2 and 3 the choice of 4 for ports 5 and 6 the choice 7 for port 8 is required to indicate that the SIP account tied to port market as Active will be used for all members of the same Hunting group Needless to say those members of the same Hunting group may not be sequential ports In following example
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