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...
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 port of SIP 60X for initial configuration or if it is being used as a router.
FXS port to be connected to analog phones / fax FXS1 - FXS8 machines Once the SIP 60X is turned on and configured, the front display panel indicates the status of the unit. Figure 2: Diagram of SIP 60X Display Panel...
SIP Server. 3 APPLICATION DESCRIPTION There are two scenarios where the SIP 60X series can be effectively used to enable any business to leverage the benefits of VoIP and the Internet. Examples of SIP 60X Configurations 3.1.1 Application : SIP 60X FXS Gateway with PBX Scenario.
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.
EN55022/EN55024 and FCC part15 Class B Safety 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)
5 BASIC OPERATIONS 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 *** on the standard analog phone connected to any FXS port.
SIP 60X User Manual Announces current Firmware Server IP Firmware Server IP address. Enter 12 digit new IP address. Address Configuration Server IP Announces current Config Server Path Address IP address. Enter 12 digit new IP address. Upgrade Protocol Upgrade protocol for firmware and configuration update.
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, 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 User Manual 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 available, use “hook flash” (toggle on-off hook quickly).
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 in a conference: 1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.
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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.
*** and 78. 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.
Also used to switch to a new channel for a new call. Pressing pound sign will serve as Re-Dial key. 7 CONFIGURATION GUIDE 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.
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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).
SIP 60X User Manual are finished. 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 make calls.
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: ...
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This is the device‟s internal HTTP server port. Default is Web Port 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...
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SIP 60X User Manual BASIC OPTIONS SETTING Setting Options Meaning Time Zone 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.
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End User Password This contains the password to access the Web Configuration Menu. This field is case sensitive. If set to “Yes”, the SIP 60X will respond to the PING Reply ICMP command from other computers, but it also is vulnerable WAN port to the DOS attack.
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PPPoE Link Up Shows whether the PPPoE connection is running if connected to DSL modem. Shows type of NAT the SIP 60X is connected to via its WAN port. It is based on STUN protocol. Port Status Shows several information regarding the individual FXS ports.
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.
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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 Lock Keypad Update via keypad is disabled.
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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...
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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.
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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.
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Reboot Allows the user to specify the time frequency (in minutes) Register for the SIP 60X to refresh its registration with the specified Expiration registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days).
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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...
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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”.
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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;...
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SIP 60X User Manual Default is Callee. This decides whether Caller or Callee Fax Tone sends out the re-INVITE for T.38 or Fax Pass Through. Detection Mode Jitter Buffer Type Select either Fixed or Adaptive based on network conditions. Jitter Buffer Length Select Low, Medium or High based on network conditions.
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SIP 60X User Manual This parameter controls 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”...
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SIP 60X User Manual Explanation of example rule (reading from left to right): ^1900x. - prevents dialing any number started with 1900 <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code...
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...
Figure 5: Screen-Shot of Rebooting Page 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.
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.
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.
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