Installation instructions for MultiVOIP 410ST/810ST software vs 5.06AK MultiVOIP's Uninstall Solution Product Update PN: 82010180 Rev. A 02/16/04 Before uninstalling any old versions of MultiVOIP software, install the new MultiVOIP Software provided for this readme. 1. Run "MultiVOIP configuration" from your old version of MultiVOIP software and take note of the current settings. Your MultiVOIP will be reset to factory defaults during this upgrade. 2. Run ISDN_BRI_506AK.exe to unzip the install files. The install files will be unzipped to the directory C:\mvp506AK. 3. Go to the folder C:\MVP506 and run setup.exe to install the software to the pc. Accept the default installation folder and program group name. When prompted, select the comm. port that you will use with the MultiVOIP. Click "No" when asked if you would like to run configuration. 4. Connect the MVP410ST/810ST to the PC comm. port and power it on. 5. Run Start->Programs->MultiVOIP 5.06->Upgrade Software. This will load factory defaults, H.323 PDL, and firmware into the MultiVOIP and will take a few minutes to complete. 6. Run Start->Programs->MultiVOIP 5.06->Configuration and configure the MultiVOIP for your application. 7. Uninstall your old version of MultiVOIP software by selecting the "Uninstall" option from the program group. 8. The upgrade is complete. Note: You will be able to manage and configure the voip with WEB interface using a Java plug in for the browser. You can retrieve vs. 1.4.02.01 from our ftp server at ftp://ftp.multitech.com/MultiVoip/Misc/java/j2re-1_4_2_01/ After installing, reboot your PC. MVP410ST/810ST Firmware History (5.06.AK) 1. Added T.38 Support in SIP Mode (faxing). 2. Proxy and SIP ports are configurable. This is an option which provides the User freedom to configure the port on knowing the remote user's port so that both of them can be configured to a common port as desired. 3. Improved H.323 and SIP interoperability with 3rd parties. 4. IP network failover PT. The purpose of this module is to have the PSTN ports busy on detecting Ethernet link is down. Whenever there is Ethernet failure the console will display 'Channel is down' after 1 minute. And again once the ethernet link is made up the console will start showing 'Channel is Up'. 5. Enabled forward on both 'Busy ' and 'No response' simultaneously. When a call is made to this channel first it will check if the specific channel is busy. If it is busy, then it will forward the call or else it will wait for the number of ring count to expire and then forward the call to the forward destination. 6. SPP fix to allow multiple VoIPs and Registrar to reside behind a NAT with a fixed IP address. We can have Registrar behind the NAT and the clients outside the NAT. These clients can register with the registrar below the Firewall. But the clients will not be able to know the actual IP address of the Registrar and still get registered with it. 7. Coder supported for eVoice call - G723, G711. This is a new feature implemented to give codec support for EVoice calls with already existing G723 codec support. The codecs which are supported now with already existing G723 are G711-A Law and G711-U Law. 8. Added local 'c' SDP support. SDP details will have Global 'C' (in SDP details which is mandatory) and Local 'C' (in media details which is optional). Earlier we always used to refer to the Global 'C' for all the media transactions. After the present changes first we will check for availability of Local 'C', if not we will be refering to the Global 'C' details. 9. Added SIP call waiting supplementary service. 10. Added SPP call waiting, call forward, call hold, and call XFER supplementary services. 11. Bug Fixes for Assisted call xfer in H.323 12. DNS Support for SIP Proxy. This support avoids complexity of remembering the IP addresses. Instead of which we will be giving Domain Name which will resolve the IP addresses which in turn used to make and receive calls. This support is also necessary to interoperate with Pingtel's SIPxchange. 13. Now validate eVoice packets before processing. 14. Added an explicit message in the console logs to print DNIS number. Currently we print only ANI number explicitly. 15. Identification of SIP call legs based on From, To and tags. 16. Added remote ping feature. 17. Added an ISDN MSN feature. MSN (Multiple Subscriber Number) in Euro-ISDN (and some country specific variants) it is possible to have several ISDN numbers for the same BRI or PRI connection. This feature allows to assign ISDN different numbers to different applications (or different physical equipment eg: Modem, Phone instrument, Fax machine etc.,). Right now, a maximum of 8 MSN numbers are supported in our configuration. For all ISDN incoming calls, when the dialed digits are matching with the configured MSN number, then if the dialNumber corresponding to the matched MSN number is present, then with that dialNumber, IP outgoing call is made. Else, for the matched MSN number, if no dialNumber is present, dial tone is provided to the user to dial the actual digits with which to make the IP outgoing call. 18. Added ISDN user configurable numbering type (calling & called party), numbering plan (called party). Support for the user to select the Calling / Called number type to be one of the following: * UNKNOWN * INTERNATIONAL * NATIONAL * NET_SPF * SUBSCRIBER * LOCAL * ABBREVIATED Support for the user to select the Called Number Plan to be one of the following: * UNKNOWN * ISDN * TELEPHONE * DATA * TELEX * NATIONAL * PRIVATE * RESERVED 19. Option for ISDN Terminal 1 to act as Terminal Clock Master. The BRI VOIP will use an internal crystal as the clock master when any one of the ports is set for Network mode. If all ports are set for Terminal mode, then the first port to activate will become the clock master and it will get its clock from the network. With this Terminal Clock Master feature, we are forcing one of the BRI VOIP ports to be in Terminal mode and be the clock master for all ports on the VOIP. When enabled, the ISDN 1 port will be the clock master for all time whether or not it has activated. 20. Fix for making successful calls when GK is not online & if we've registered with GK. When "Register with Gatekeeper" option was enabled in the configuration & if the GK is not online / not reachable, we were unable to make calls from Gateway, though a valid Outbound PhoneBook entry was present for the dialed number. 21. Fix for "Autocall enabled on channel 2" without any autocall number, after downloading default cnf file. Whenever 5.05 default cnf was downloaded, by default, autocall was enabled on channel 2 without any autocall number. This bug was fixed. 22. Fix for H.323 - Alternate routing. H.323 Alternate Routing feature where if the primary IP Address configured for IP outgoing call is congested / not reachable, it was not switching to the alternate IP Address configured. 23. Fix for SPP Alternate Routing. SPP Alternate Routing feature where if the primary Phone number configured for IP outgoing call is congested / not reachable, it was not switching to the alternate Phone number configured. 24. Fix for sending dialed DTMF digits thru Q.931 INFO messages. The Dialed digits were never passed from the MVP410ST/810ST to the PBX in an INFO message and thus the PBX never stopped playing the dial tone because of TrunkId, ChannelId not matching properly. 25. Fixes for Point-To-Point & Point-To-MultiPoint Network Interface type configuration & functionality. Though we allow the user to configure 8 TEIs if the interface type configured is Network, we set Point-to-Point / Point-to-MultiPoint configuration in ISDN stack based on first TEI value. So modified the configuration to display only 1 TEI or 2 TEIs (if operator selected is USA), for the user to configure. 26. Fix for ringback problem when a call is made from NGAV to BRI. When an IP call is made from NGAV to BRI, there was no ringback heard on NGAV.