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Read moreabout the condition Brand: Grandstream Model: Grandstream IP PBX Solutions Interface: Ethernet (RJ-45) MPN: Grandstream IP PBX Solutions Telecom Technology Equipment Visit my eBay store Sign up for newsletter Search Store iPong Pro iPong Topspin HOME & PERSONAL SECURITY SETS Store Categories Store home AudiocodesCONTAINER SETSCutting Edge SolutionsGigasetGrandstreamHOBBY & COLLECTOR SETSHOME & PERSONAL SECURITY SETSHtekIP SolutionsMagic ChefOutdoor EdgePanasonicPlantronicsPolycomSnomTABLE TENNIS SETSUnidenValcomVTechYealinkYeastarOther ALSO GET 5% OFF YOUR TOTAL WHEN YOU PURCHASE FROM THIS LISTING [ Let TTE help you purchase your VoIP SIP solution. Today, via a single listing, you can chose the right IP PBX VoIP elements to complete you IP technology package. Stop searching multiple sellers on multiple market places. Find answers to all your questions with our FAQ section of this listing. Contact TTE to get the latest firmware, installation guides, brochures, datasheets and more for all our Grandstream products. Contact TTE via eBay messaging for directions on Grandstream product provisioning, trouble shooting, interoperability, and product related questions. Feel at ease with our 14-day money back guarantee and 1-year manufacturer's warranty. Excellent quality and customers service is our #1 goal. Due to text limitations, You can find details for each product sold in this listing on Grandstream's website or you can Contact me directly for details like datasheets, product descriptions, troubleshooting, etc. My contact information is on my storefront page. ] GS-UCM6510 UCM6510 innovative IP PBX appliance by Grandstream UPC: 6947273701651SKU: UCM6510 innovative IP PBX applianceTTE P/N: GS-UCM6510 1GHz quad-core Cortex A9 application processor, large memory (1GB DDR3 RAM, 32GB Flash) and dedicated high performance multi-core DSP array for advanced voice processing 1 integrated T1/E1/J1 interface, 2 PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with lifeline capability in case of power outage and up to 50 SIP trunk accounts Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based caller ID/call progress tone and smart automated impendance matching for various countries Gigabit network port(s) with integrated PoE, USB, SD card; integrated NAT router with advanced QoS support GS-UCM6116 UCM6116 innovative IP PBX appliance by Grandstream UPC: 6947273701279SKU: UCM6116 innovative IP PBX applianceTTE P/N: GS-UCM61161GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing Integrated 16 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in case of power outage, and up to 50 SIP trunk accounts Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based caller ID/call progress tone and smart automated impendance matching for various countries Supports up to 500 SIP endpoint registrations, up to 60 concurrent calls (up to 40 SRTP encrypted concurrent calls), and up to 32 conference attendees Flexible dial plan, call routing, site peering, call recording, central control panel for endpoints, integrated NTP server, and integrated LDAP contact directory Automated detection and provisioning of IP phones, video phones, ATAs, gateways, SIP cameras, and other endpoints for easy deployment Strongest-possible security protection using SRTP, TLS, and HTTPS with hardware encryption accelerator GS-UCM6108 UCM6108 innovative IP PBX appliance by Grandstream UPC: 6947273701262SKU: UCM6108 innovative IP PBX applianceTTE P/N: GS-UCM6108 1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing Integrated 8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in case of power outage, and up to 50 SIP trunk accounts Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based caller ID/call progress tone and smart automated impendance matching for various countries Supports up to 500 SIP endpoint registrations, up to 60 concurrent calls (up to 40 SRTP encrypted concurrent calls), and up to 32 conference attendees Flexible dial plan, call routing, site peering, call recording, central control panel for endpoints, integrated NTP server, and integrated LDAP contact directory Automated detection and provisioning of IP phones, video phones, ATAs, gateways, SIP cameras, and other endpoints for easy deployment Strongest-possible security protection using SRTP, TLS, and HTTPS with hardware encryption accelerator GS-UCM6104 UCM6104 innovative IP PBX appliance by Grandstream UPC: 6947273701255SKU: UCM6104 innovative IP PBX applianceTTE P/N: GS-UCM6104 1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing Integrated 4 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in case of power outage, and up to 50 SIP trunk accounts Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based caller ID/call progress tone and smart automated impendance matching for various countries Supports up to 45 concurrent calls Flexible dial plan, call routing, site peering, call recording, central control panel for endpoints, integrated NTP server, and integrated LDAP contact directory Automated detection and provisioning of IP phones, video phones, ATAs, gateways, SIP cameras, and other endpoints for easy deployment Strongest-possible security protection using SRTP, TLS, and HTTPS with hardware encryption accelerator GS-UCM6102 UCM6102 innovative IP PBX app. by Grandstream UPC: 6947273701248SKU: UCM6102 innovative IP PBX app.TTE P/N: GS-UCM6102 GS-UCM6208 UCM6208 IP PBX 8FXO, 2FXS Appliance by Grandstream UPC: 6947273702139SKU: UCM6208 IP PBX 8FXO, 2FXS ApplianceTTE P/N: GS-UCM6208 SMB IP PBX with 8 FXO and 2 FXS PortsSupports up to 800 users, 50 SIP trunks, and up to 100 concurrent callsZero configuration provisioning of Grandstream SIP endpointsSRTP, TLS, and HTTPS encryptionDual Gigabit network ports with integrated PoESupports up to a 5-level IVRBuilt-in call recording server; access recordings via web user interfaceSupports call queueBuilt-in Call Detail Recordings (CDR) for tracking phone usage by line, date, etc.Multi-language auto attendantIntegrated LDAP and XML phonebooksSupports any SIP video endpoint that uses the H.264, H.263, or H.263+ codecsSupports voicemail and fax forwarding to emailUSB and SD portsUp to 6 password protected conference bridges that allow up to 32 simultaneous PSTN or IP participants GS-UCM6204 UCM6204 IP PBX 4FXO, 2FXS Appliance by Grandstream UPC: 6947273702122SKU: UCM6204 IP PBX 4FXO, 2FXS ApplianceTTE P/N: GS-UCM6204 SMB IP PBX with 4 FXO and 2 FXS PortsSupports up to 500 users, 50 SIP trunks, and up to 45 concurrent callsZero configuration provisioning of Grandstream SIP endpointsSRTP, TLS, and HTTPS encryptionDual Gigabit network ports with integrated PoESupports up to a 5-level IVRBuilt-in call recording server; access recordings via web user interfaceSupports call queueBuilt-in Call Detail Recordings (CDR) for tracking phone usage by line, date, etc.Multi-language auto attendantIntegrated LDAP and XML phonebooksSupports any SIP video endpoint that uses the H.264, H.263, or H.263+ codecsSupports voicemail and fax forwarding to emailUSB and SD portsUp to 3 password protected conference bridges that allow up to 25 simultaneous PSTN or IP participants FAQs: Can I invite other parties to join the conference room? Yes. From UCM6100 series web GUI->PBX->Call Features->Conference, there is an option icon "Invite a participant" on each conference room. Click on it and enter the extension number of the party you would like to invite. Then click on "Add". A call will be sent to the extension to participate in the conference. A conference participant can also invite other parties to the conference from the phone during the call. Please make sure option "Enable User Invite" is turned on for the conference room first. Then you can enter 0 (with participant's permission) or enter 1 (without participant's permission) during the conference call, and follow the voice prompt to invite other party to the conference. Note: Conference administrator can always invite other parties from the phone during the call, by entering 0 (with participant's permission) or 1 (without participant's permission). To join a conference room as administrator, enter the admin password when joining the conference.. Can I restore a backup file from one UCM model on another different UCM model? Yes, starting from firmware 1.0.11.27, backup files are compatible among UCM61xx/UCM62xx models. Users can restore backup files from one UCM61xx/UCM62xx model to another. However, if you are restoring a backup file from a higher UCM model number to a lower one, you will be warned and prompted that the additional feature (e.g., FXO ports, extensions or conference rooms) is not available and the restore will not be allowed. This happens when the backup file is from a model that has more FXO ports, extensions or conference rooms configured than the maximum supported capability on the target UCM. For example, when the user is trying to restore a backup file from UCM6108 on UCM6104, if FXO port 5 to 8 are configured on UCM6108, the user will not be able to restore the backup file on UCM6104 because UCM6104 only supports FXO port 1 to 4. In this case, please manually delete the configuration of FXO port 5 to 8 and create backup on UCM6108 again to be restored on UCM6104. Can I restore and backup the UCM6100 series configurations? Yes. The UCM6100 series supports local backup and network backup under web GUI->Maintenance->Backup. - Local backup. Select the files to be backed up locally. After backup, a list of previous backup files will be displayed in this page. The backup files can be downloaded or directly used for restore. Please note that backing up voice, voicemail, voice record and CDR files requires SD card or USB Flash drive to be plugged into the UCM6100 series first. - Network backup. Configure the SFTP server login information and backup time to and backup the UCM6100 series files remotely to a SFTP server. Additionally, users can go to web GUI->PBX->Maintenance->Cleaner to set up schedule and clean up the CDR as well as voice records in UCM6100 series after backup. Can we connect external USB hard drive to UCM6100/UCM6510 via USB port? What are the supported format and what is the max capacity allowed for the external USB hard drive? UCM6100/UCM6510 support format NTFS, FAT32 and EXT3/EXT4 for the connected external USB hard drive. Theoretically, UCM6100/UCM6510 do not have size limitation for the connected USB external hard drive. The supported capacity is the same as when the external USB hard drive is connected to a PC. However, there is limitation for the power supply on the UCM6100/UCM6510 USB port. If the external USB hard drive doesn't have external power supply, it might not get properly powered up thus it will not work after connecting to the UCM6100/UCM6510 USB port. It is recommended to use the external USB hard drive with external power supply. Do all the UCM6100 series models support PoE? Does the FXS port work under PoE? Yes, all UCM6100 series models support PoE. All the interfaces and functions (including FXS port) work under PoE. Does the UCM6100 series support blacklist? Yes, the UCM6100 series supports blacklist for trunks. It can be configured under UCM6100 series web GUI->PBX->Basic/Call Routes- >Inbound Routes->Blacklist. The blacklist (by CallerID) is used for all inbound routes. Does the UCM6100 series support call recording? How does it work? Yes, the UCM6100 series supports call recording, for extensions as well as conference room. For an extension to record an ongoing call, go to UCM6100 series web GUI->PBX->Internal Options->Feature codes to enable and configure the feature code for "Audio Mix Record" first. Then dial the feature code during the call. The recorded file can be downloaded in the corresponding CDR report entry under web GUI->Status->CDR. To record conference calls, select the particular conference room to edit and turn on the "Record Conference" option first. The conference calls will then be automatically recorded. The recording file can be viewed and downloaded under PBX->Call Features->Conference. "Auto Record" option is also supported per extension, analog trunk, VoIP trunk, ring group and call queue. Once enabled from the web UI configuration, the calls going through the module will be automatically recorded. Does the UCM6100 series support Fax? Yes. UCM6100 series supports T.30/T.38 Fax and Fax Pass-through. After receiving the Fax, UCM6100 series can convert it to PDF format and send it to the configured Email address. To do this, users could turn on "Fax Detection" for a specific VoIP trunk under UCM6100 series web GUI->PBX->Basic/Call Routes->VoIP Trunks. Or users can set up the extension for Fax (under web GUI->PBX->Internal Options->Fax/T.38) and then configure PBX->Call Features->IVR->Key Pressing Events to have the key pressing event to go to the extension of the Fax. Does the UCM6100 series support SIP video call? How many concurrent video calls are supported? The UCM6100 series supports SIP video call between SIP video phones. Currently, UCM6100 series supports static payload type only, i.e., the payload type needs to be 99 for H.264. When using H.264 and QVGA with bit rate 256kbps, the UCM6100 series supports 30 concurrent video calls (approximately). When using H.264 and QVGA with bit rate 512kbps, the UCM6100 series supports 25 concurrent video calls (approximately). Note: UCM6100 series doesn't support video conference. Does the UCM6100 series support video voicemail? The UCM6100 series doesn't support video voicemail as it doesn't do video transcoding. The video codecs are still listed in SIP trunk configurations as it needs to be configured so that it can be used in SDP exchange. How can I configure the UCM6100 series to manage the permission and privilege for outbound calls? Outbound rules can be configured in UCM6100 series web GUI->PBX->Basic/Call Routes->Outbound Rules. Users could manipulate the following options to manage the permission and privilege for outbound calls: - "Password" option for outbound rule. This is the password required to use the outbound rule when making outbound calls from the UCM6100 series extensions. - "Skip Trunk Auth" option for extension. This can be configured per extension under web GUI->PBX->Basic/Call Routes->Extensions. When enabled, this extension does not need to enter the Password as required to use the outbound rule. - "Permission" option for extension and "Privilege Level" option for outbound rule. Users could assign "Internal", "Local", "National" or "International" to extension's "Permission" and outbound rule's "Privilege Level". The extension needs to have higher level permission than the outbound rule's privilege level to make outbound calls using this rule. -"Enable Filter on Source Caller ID". If enabled, "Privilege Level" will be automatically disabled. Only the listed extensions or the extension matching the “Custom Dynamic Route” will be allowed to use the outbound route.. How can I manage the network security on the UCM6100 series? The UCM6100 series provides Firewall settings under web GUI->Settings->Firewall. Typical options such as "Ping Defense", "Syn-Flood Defense" and "Ping-or-Death Defense" can be configured under "Static Defense" page; Blacklist is supported to prevent massive TCP connections from malicious hosts under "Dynamic Defense" page. The UCM6100 also has a Fail2Ban defensive mechanism under web GUI->Settings->Firewall->Fail2Ban. For more security details, please refer to the UCM security that you can find in our resource library How can I resolve FXO/PSTN issues (such as echo issue, the call doesn't hang up, and etc) on the UCM6100 series? Go to the UCM6100 series web GUI->PBX->Basic/Call Routes->Analog Trunks. When creating or editing the analog trunk, there is an option "PSTN Detection" for users to automatically detect FXO/PSTN parameters (Busy tone, Polarity Reversal and Current Disconnect). Please make sure there are more than one channel configured and working for PSTN detection. Users could also detect the best ACIM settings for each FXO port by using "ACIM Detection" in "FXO Ports" under web GUI->PBX->Internal Options->Hardware Config. How can I setup the UCM6100 series to auto detect fax tones and send the fax as PDF? 1. Create a Fax extension on the UCM6100 series under Internal options FAX/T.38. 2. Under Fax Setting enter your email address in the Default email address. 3. Enable Error correction mode in the Fax setting. 4. Under Call Features on the IVR menu, create an IVR and enable options Dial other extensions and Dial trunk. Note: With these settings the IVR will auto-detect the Fax tones and send the fax to the default email address. Please check your spam folder for the fax email. How can I troubleshoot the UCM6100 if it cannot detect hang-up on the FXO port? If the UCM6100 cannot detect hang-ups on the FXO port, the FXO port will not be released. This might cause the line always busy, or long voicemail message sent to the extension if the outside caller leaves a voicemail. Please follow the instructions below to troubleshoot the issue. 1. Go to the UCM6100 web GUI->PBX->Analog Trunks page. 2. Select the analog trunk configured for the FXO port. 3. Run "PSTN Detection" by clicking on "Detect" button. This will correct the disconnect method for the analog trunk in order for the UCM6100 to detect the hang-up. If the issue still persists, users might need check whether the PSTN disconnect tone's energy is high enough to trigger the disconnect detection as configured on the UCM6100's analog trunk. The disconnect threshold value on the UCM6100 can be found in "Busy Tone" and "Congestion Tone" settings. If the "Tone Country" is set to "United States of America (USA)", the busy tone and congestion tone are set as follows: Busy Tone: f1=480@-50,f2=620@-50,c=500/500 Congestion Tone: f1=480@-50,f2=620@-50,c=250/250 where "-50" is the threshold value to detect the hang-up. The actual energy (in dB) for threshold "-50" can be calculated as below: (-50) x 0.1 - 23 = -28 dB This means the PSTN disconnect tone's energy has to be higher than -28dB in order for the UCM6100 to detect the hang-up. If it's lower than -28dB, on the UCM6100 analog trunk, users will need set the "Tone Country" to "Custom", calculate the threshold value based on the actual PSTN disconnect tone's energy, and change the settings for the "Busy Tone" or "Congestion Tone" in order to meet the disconnect requirement. Note: To decide whether the PSTN line is using busy tone or congestion tone to hang up the call, users can do a quick test as described below. 1. Make a phone call from outside PSTN line to a UCM6100 extension and establish the call. 2. Hang up the call from the outsider caller. 3. Listen to the tone on the UCM6100 extension side. If users immediately hear busy tone, please configure the threshold for "Busy Tone". If users hear dial tone first and then busy tone, please configure the threshold for "Congestion Tone". For example, the actual PSTN disconnect tone is congestion tone and has energy value -33dB, users might change the congestion tone setting to the following: F1=480@-130,f2=620@-130,c=250/250 How can I troubleshoot the UCM6100 series? Is there any log I can get or any tool I can use on the UCM6100 series? Yes, the UCM6100 series provides troubleshooting interface under web GUI->Maintenance. - In web UI->Maintenance->Syslog page, fill out the syslog server, select modules and set up syslog level. The syslog information will be printed out to the syslog server as configured. - In web UI->Maintenance->Troubleshooting page, users can capture trace from UCM6100 network interface in "Ethernet Capture", or use the "IP Ping" and "Traceroute" to track network related issues. - In web UI->Maintenance->Troubleshooting->Analog Record Trace page, users can generate record file for analog trunk calls to troubleshoot issues related to PSTN calls. How can I upgrade the UCM6100 series via remote HTTP/HTTPS/TFTP server? The UCM6100 series can be upgraded via remote HTTP/HTTPS/TFTP server besides local uploading. To upgrade via remote HTTP/HTTPS/TFTP server, fill in the firmware server path and upgrading method in web GUI->Maintenance->Upgrade. Save the change and reboot the UCM6100 series to initiate the upgrading process. During the upgrading process, the UCM6100 series service will still be running. A manual reboot message "Upgrade Result: Succeeded. Pls Reboot" will prompt on UCM6100 series LCD once the upgrading is finished, which is to avoid immediate service interruption from automatic reboot. Please manually reboot the UCM6100 series from web GUI or power cycle it when it's appropriate. The UCM6100 series will then boot up with the new firmware version. How can I use LDAP in UCM6100 series? The UCM6100 series has a built-in LDAP server for users to manage corporate phonebook. By default, the LDAP server has generated a phonebook based on the extensions created. If users have the Grandstream phone provisioned by the UCM6100 series, the LDAP directory will be automatically set up on the phone side and can be used right away. To manage the LDAP server configuration and add more phonebook information, please go to web GUI->Settings->LDAP Server. How do I setup a remote extension to the UCM6100 series IPPBX? If the UCM is using an external IP (not behind a NAT), then you don’t need to configure anything for remote extensions, but if it is behind a NAT then these are the steps: - Navigate to PBX-->SIP Settings-->NAT on the UCM6100's web UI. Put the external IP of the network in the field “External IP Address” (if a domain is being used instead. e.g. DDNS, then use the next field “External Host”) and the internal IP of the UCM in the field “Local Network Address” - Port forward in your router the SIP port for the UCM (by default UDP:5060 and can be changed under the “PBX” > “SIP Settings” > “General” tab –we recommend changing it to increase security) and the audio ports (by default the range UDP:10000-20000 and can be changed/decrease under the “PBX” > “Internal Options” > “RTP Settings” tab – we recommend decreasing it to 10000-11000). - On the remote phone(s) use the external/public IP of the UCM as the SIP server. Also put in the SIP User ID, Authenticate ID, SIP Password for the remote extension. - On the remote phones you may want to enable “Keep-Alive” for NAT settings. In Grandstream phones the option is “Auto” for the setting “NAT Traversal” located under the “Accounts” > “Accounts #” > “Network Settings” tab of the phone’s Web interface. If those options do not work then select “STUN” and put “stun.ipvideotalk.com” in the field “STUN Server” located under the “Settings” > “General Settings” tab also of the phone’s Web interface. - On the phone(s) you may want to enable “Use Random Port” by setting it to “Yes” under the “Settings” > “General Settings” tab of the phone’s Web interface. For this setting you need to reboot the phone to take effect. How do I use Fail2Ban on the UCM6100? The UCM6100 supports Fail2Ban for authentication errors (SIP REGISTER/INVITE/SUBSCRIBE) to help secure the PBX system. To enable Fail2Ban, please log in the UCM6100 web GUI and go to Settings->Firewall->Fail2ban page. Make sure you have enabled both "Enable Fail2Ban" under Global Settings and "Asterisk Service" under Local Settings first. Then fill out the following parameters in the web page to have Fail2Ban work as desired on the UCM6100. Global Settings: - Banned Duration: Configure the duration (in seconds) for the detected host to be banned. The default setting is 300. If set to -1, the host will be always banned. - Max Retry Duration: Within this duration (in seconds), if a host exceeds the max times of retry as defined in "MaxRetry", the host will be banned. - MaxRetry: Configure the number of authentication failures during "Max Retry Duration" before the host is banned. - Fail2Ban Whitelist: Add host address in the whitelist. It could be IP address, CIDR mask or DNS host. Fail2Ban will not ban the host with matching address in this list. Up to 5 addresses can be added into the whitelist. Local Settings: - Port: 5060 only. - MaxRetry: Configure the number of SIP authentication failures during "Max Retry Duration" before the host is banned. Please make sure this option is properly configured as it will override the "MaxRetry" value under "Global Settings". How many agents can be added in one call queue on the UCM6100 series? The UCM6100 series supports up to 100 agents to be added in one call queue. How many BLF extensions are supported on the UCM6100 series? On the UCM6100 series, there is no specific limitation on the number of BLF extensions. However, the PBX performance can be affected by the number of subscribers and how the BLF feature is used. For example, 300 subscribers to 50 BLFs will generate much heavier impact than 10 subscribers to 150 BLFs. Also, if the event list BLF, especially the remote BLF monitoring is used, please be aware that adding all the PBX extensions for every office with a peer UCM6100 to the event list could severely impact the UCM6100 performance because of unnecessary state changes sent to every subscriber. How many call queues does the UCM6100 series support? The UCM6100 series supports up to 10 call queues. How many concurrent calls does the UCM6100 series support? UCM6102 supports up to 30 simultaneous calls; UCM6104 supports up to 45 simultaneous calls; UCM6108/6116 supports up to 60 simultaneous calls. Note: The internal extension-to-extension calls are included as concurrent calls. How many conference bridges can be used simultaneously on the UCM6100 series? UCM6102/6104 supports up to 3 conference bridges allowing up to 25 simultaneous PSTN or IP participants. UCM6108/6116 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP participants. How many extensions are supported on the UCM6100 series? The UCM6100 series supports up to 500 extensions. How many members can be added in one ring group on the UCM6100 series? The UCM6100 series supports up to 100 agents to be added in one ring group. How to allow users to dial trunk provider feature codes? The UCM6100 has built in feature codes, so there are times where UCM feature codes conflict with the trunk provider feature codes. The way around this would be to create an outbound pattern that essentially dials the providers feature code. For example, *72 on the UCM is used for "Unconditional Call Forwarding", but the trunk provider has their *72 set as "Do Not Disturb". If the user attempts to enable "Do Not Disturb" by dialing *72, they would be setting "Unconditional Call Forwarding" instead. Here's how to get around this: - Navigate to Web UI -> PBX -> Basic/Call Routes -> Outbound Routes - Click on "Create New Outbound Rule" - Calling Rule Name is for reference purposes. In this example it would be set as "TrunkDND" - Pattern: Configure a pattern that does not conflict with the current feature codes. E.g. 88 - Use Trunk: Select the trunk that the feature code is needed - Strip: Before appending the trunk feature code, 88 must be striped from the dialed string. In this case 2 digits will be stripped - Append: After stripping the 88 from the dialed string append the trunk provider's "Do Not Disturb" feature code. In this case it would be *72 - Click "Save" at the bottom then "Apply Changes" at the top. For this example, if a user dials "88" it will be routed through the specified trunk, strip "88", append "*72" then send the manipulated string through the trunk to enable DND. My application with UCM6100/6510 AMI was working on UCM6100/6510 firmware version 1.0.9.26, but it doesn't work anymore after it's upgraded to 1.0.10.39. What should I do? UCM6100/6510 firmware 1.0.9.26 is using Asterisk 1.8 which has AMI supported. UCM6100/6510 firmware 1.0.10.39 has upgraded to Asterisk 13 which is using AMI v2 (starting in Asterisk 12, the AMI is upgraded to AMI v2). According to Asterisk Wiki https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12, on AMI v2, many events were changed, and the semantics of channels and bridges were defined. In particular, how channels and bridges behave under transfer scenarios and situations involving multiple parties has changed significantly. Please see the AMI v2 Specification for more information: https://wiki.asterisk.org/wiki/display/AST/AMI+v2+SpecificationSince AMI is provided as-is, please consult with the software vendor to update your application using AMI v2. Or if you wish to downgrade UCM6100/6510 from 1.0.10.39 to 1.0.9.26, please contact our technical support for information and instructions. Should I always enable all Syslog on the UCM6100 in case troubleshooting is needed? It’s not recommended to open all levels to all the syslog modules. Too many syslog print might cause traffic and affect system performance. GS-UCM6202 UCM6202 IP PBX 2FXO, 2FXS Appliance by Grandstream UPC: 6947273702115SKU: UCM6202 IP PBX 2FXO, 2FXS ApplianceTTE P/N: GS-UCM6202 SMB IP PBX with 2 FXO and 2 FXS PortsSupports up to 500 users, 50 SIP trunks, and up to 30 concurrent callsZero configuration provisioning of Grandstream SIP endpointsSRTP, TLS, and HTTPS encryptionDual Gigabit network ports with integrated PoESupports up to a 5-level IVRBuilt-in call recording server; access recordings via web user interfaceSupports call queueBuilt-in Call Detail Recordings (CDR) for tracking phone usage by line, date, etc.Multi-language auto attendantIntegrated LDAP and XML phonebooksSupports any SIP video endpoint that uses the H.264, H.263, or H.263+ codecsSupports voicemail and fax forwarding to emailUSB and SD portsUp to 3 password protected conference bridges that allow up to 25 simultaneous PSTN or IP participants ALSO GET 5% OFF YOUR TOTAL WHEN YOU PURCHASE FROM THIS LISTING Questions and answers about this item No questions or answers have been posted about this item. Ask a question - opens in a new window or tab Seller assumes all responsibility for this listing. Shipping and handling This item will ship to Thailand, but the seller has not specified shipping options. Contact the seller- opens in a new window or tab and request a shipping method to your location. Shipping cost cannot be calculated. Please enter a valid ZIP Code. Item location: Austin, Texas, United States Shipping to: Americas, Europe, Asia, Australia, South Africa, New Zealand No additional import charges at delivery! This item will be shipped through the Global Shipping Program and includes international tracking. 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Please enter a number less than or equal to 24. Select a valid country. ZIP Code: Please enter a valid ZIP Code. Please enter 5 or 9 numbers for the ZIP Code. Shipping and handling Import charges (estimated) To Service Delivery* US $33.11 $42.17 Thailand International Priority Shipping Estimated between Wed. Dec. 21 and Wed. Dec. 28 * Estimated delivery dates- opens in a new window or tab include seller's handling time, origin ZIP Code, destination ZIP Code and time of acceptance and will depend on shipping service selected and receipt of cleared payment- opens in a new window or tab. Delivery times may vary, especially during peak periods. Domestic handling time Estimated sales tax Will usually ship within same business day if paid before 12:00 PST (excludes weekends and holidays). Expected ship time may vary and is based on seller's order cut-off time. Seller charges sales tax in multiple states. Return policy After receiving the item, contact seller within Refund will be given as Return shipping Restocking Fee 14 days Money back or item exchange (buyer's choice) Buyer pays return shipping 20% restocking fee may apply Return policy details 14-Day Money back or exchange on DAMAGED or DEFECTIVE goods ONLY. I can't accept returns on items because you changed your mind about an item. Payment details Payment method Preferred / Accepted PayPal Preferred Immediate payment required for this item Immediate payment of US $519.00 is required.