Frequently asked Questions Troubleshooting

Transcription

1 VoIP Telephones Frequently asked Questions Troubleshooting FernTel IP Family (Ex)ResistTel IP2 Family FHF FAQ-TS-IP_Phones_EN 02/17 V10

2 FAQ-TS-IP_Phones_EN 02/17 V10 Frequently asked Questions / Troubleshooting Brand names are used with no guarantee that they may be freely employed. Almost all hardware and software designations in this manual are registered trademarks or should be treated as such. All rights reserved. No part of this manual may be reproduced in any way (print, photocopy, microfilm or by any other means) or processed, duplicated or distributed using electronic systems without explicit approval. Texts and illustrations have been compiled and software created with the utmost care, however errors cannot be completely ruled out. This documentation is therefore supplied under exclusion of any liability or warranty of suitability for specific purposes. FHF reserves the right to improve or modify this documentation without prior notice. Please read the operating manual carefully before installing a device. Copyright 2017 FHF Funke + Huster Fernsig GmbH Gewerbeallee Mülheim an der Ruhr Tel +49 (208) Fax +49 (208) Page 2 Frequently asked Questions / Trouble Shooting

3 Table of Contents 1 Frequently asked Questions Default User Name and Password FernTel IP, Default User Name and Password ResistTel IP2, Default User Name and Password ExResistTel IP2, Default User Name and Password Structure of the Software Version Information Firmware Format (Hotfix Release) Firmware Format (Development Release) Bootcode Format (Hotfix Release) Bootcode Format (Development Release) Software Update Software Update under Retention of the Main Version of the Software Software Update from Version V6 to Version V7 or higher Software Update from Version V7 to a higher Version (Ex)ResistTel IP2 (INMETRO, EAC) FernTel IP (ZB, Z2 (INMETRO, UL)) Update from Version V7 to Version V8 or V Update from Version V7, V8 or V9 to Version V10 or higher Bootcode and Firmware Update Setup of the Language of the User Interface of the Telephone Setup of the Language via DHCP Server Setup of the Language at the Telephone Setup of the Language with the Web Interface Troubleshooting Commissioning Registration Port Number, Protocols SIP, TSIP and SIPS Making Calls SIP Accounts Specifics of PBX s CX (Software PBX) Avaya Call Manager Block Dialing Licensing Protocols H SIP Frequently asked Questions / Trouble Shooting Page 3

4 4.3 Cisco Call Manager Express Block Dialing Unified Communications Manager (Call Manager) Block Dialing Configuring Unified Communications Manager for Non- Cisco SIP Phones How Cisco Unified Communications Manager identifies a Third-Party SIP Phone Third-Party SIP Phones and TFTP Enabling Digest Authentication for Third-Party SIP Phones Third-Party SIP Phone Configuration Checklist DTMF Reception Licensing Third-Party SIP Phones Where to find more Information NEC SV General Characteristics Port Numbers for the Registration Supplementary Services and Limitations Deleting the Telephone Type in the PBX Setup for FHF Telephones (FernTel IP, (Ex)ResistTel IP2) General Protocol H Protocol SIP Software Version V Software Version V Software Version V10 or higher Changing the Protocol Starting a Conference from a FHF Telephone Call Forwarding Power over Ethernet (PoE) FernTel IP Family ResistTel IP2 Family ExResistTel IP2 (INMETRO, EAC) Family Ethernet Connection over a greater Distance Ethernet Extender WLAN Optical Ethernet (Fibre Optic Cable) External Signalling Devices Page 4 Frequently asked Questions / Trouble Shooting

5 7.1 Connection of the weatherproof Signalling Device TWIN-Kombi / Type 5843/5 to weatherproof VoIP Telephones Setup of the Ringing Volume Setup of the Operating Mode Setup of the Ringing Signal Setup of the Telephone FernTel IP Family ResistTel IP2 Family ExResistTel IP2 (INMETRO, EAC) Family Connection of the explosion proof TWIN-EExII Type 5842/1 to a VoIP Telephone Setup of the DIP switch (Operating Mode / Melody) Terminal Connections Wiring Setup of the Telephone ExResistTel IP2 (INMETRO, EAC) Family Device internal Troubleshooting Tracing Wireshark General RPCAP Server and Microsoft Windows Configuration of the RPCAP Server Capturing with the Program Wireshark Older Wireshark Versions Newer Wireshark Versions Utility Programs SoftPerfect Network Scanner Tera Term Professional TFTPD Known Problems Service cannot be started Problems storing the Configuration OpenLDAP Linphone H323Phone WirelessNetView Additional IP Address at an Ethernet Interface Abbreviations Notes Frequently asked Questions / Trouble Shooting Page 5

6 1 Frequently asked Questions 1.1 Default User Name and Password In order to set up the VoIP telephone via the Web interface, you must login with a user name and a password. The information concerning default user name and default password can be seen in the manual of each device. This information is also part of the following sub chapters. User name and password can be changed in the menu General > Admin of the Web interface. The same user name and the same password will be checked changing the configuration at the interface of the phone, if the option "Protect Configuration at Phone" is active or if someone is connecting to the telephone using the protocol telnet. Page 6 Frequently asked Questions / Trouble Shooting

7 1.1.1 FernTel IP, Default User Name and Password In initial condition of a VoIP telephone of the FernTel IP family, the user name is admin and the password ip ResistTel IP2, Default User Name and Password In initial condition of a VoIP telephone of the weatherproof ResistTel IP2 family, the user name is admin and the password ip ExResistTel IP2, Default User Name and Password In initial condition of a VoIP telephone of the ExResistTel IP2 family, the user name is admin and the password ip Structure of the Software Version Information Firmware Format (Hotfix Release) Old firmware format (hotfix release): [yy-mmhhh.rr] yy mm hhh rr year information main version of the software main release release New firmware format (hotfix release): [mmhhh.rr] mm hhh rr main version of the software main release release Firmware V7: Since version [70600] there are hotfix releases only (structure see development releases). Firmware V8: Since version [80796] there are hotfix releases only (structure see development releases). Firmware V9: Since version [ ] there are hotfix releases only. Frequently asked Questions / Trouble Shooting Page 7

8 New firmware format V9 (hotfix release): [m.hhhrrr] m hhh rrr main version of the software main release release Firmware Format (Development Release) Old firmware format (development release): [yy-mmsss] yy mm sss year information main version of the software release New firmware format (development release): [mmsss] mm sss main version of the software release Bootcode Format (Hotfix Release) Bootcode format software V6: [bbb] bbb version of the bootcode software Old bootcode format software V7 and higher (hotfix release): [yy-mmhhhrr] yy mm hhh rr year information main version of the software main release release New bootcode format software V7 and higher (hotfix release): [mmhhhrr] mm hhh rr main version of the software main release release Page 8 Frequently asked Questions / Trouble Shooting

9 Bootcode V7: Since version [70600] there are hotfix releases only (structure see development releases). Bootcode V8: Since version [80796] there are hotfix releases only (structure see development releases). Bootcode V9: Since version [ ] there are hotfix releases only. New bootcode format V9 (hotfix release): [mhhhrrr] m hhh rrr main version of the software main release release Bootcode Format (Development Release) Old bootcode format software V7 and higher (development release): [yy-mmsss] yy mm sss year information main version of the software release New bootcode format software V7 and higher (development release): [mmsss] mm sss main version of the software release 1.3 Software Update Software Update under Retention of the Main Version of the Software The description is available in the manual. Frequently asked Questions / Trouble Shooting Page 9

10 1.3.2 Software Update from Version V6 to Version V7 or higher 1. Update the firmware to the hotfix version [ ] or a higher V6 release. 2. Reset the device. 3. Update the bootcode to version [225] or a higher V6 version. 4. Switch the power off and on again. 5. Update the firmware to the version V7 or a higher Version. 6. Reset the device. 7. Update the bootcodes to the version V7 or a higher version. 8. Reset the device. 9. Update the Firmware to version V7 or a higher version again. This is necessary to activate the new checksum calculation for the firmware. 10. Reset the configuration to factory default Software Update from Version V7 to a higher Version (Ex)ResistTel IP2 (INMETRO, EAC) The description is available in the manual FernTel IP (ZB, Z2 (INMETRO, UL)) Update from Version V7 to Version V8 or V9 The description is available in the manual. Page 10 Frequently asked Questions / Trouble Shooting

11 Update from Version V7, V8 or V9 to Version V10 or higher For the software versions V6, V7, V8 und V9 the telephone FernTel IP (ZB, Z2 (INMETRO, UL)) must be implemented with 8 MB DRAM and 4 MB Flash devices (Hardware ID 101 or 102). Since the software version V10 the telephone must be implemented with 16 MB DRAM and 8 MB Flash devices (Hardware ID 601 or 602), if the firmware with all protocol variants should be loaded. The telephones with the extended memory appear in the presentation of the web page with the device type: FernTel IP A. The telephone FernTel IP A can be loaded via software update with another V10 release or a higher version only. For uploading the firmware the file ip150a.bin and for uploading the bootcode the file boot150a.bin will be necessary. For upgrading the old telephones with a small memory to a software version V10 or higher, there are three different firmware variants, each missing some protocol variants. File Description boot150.bin Bootcode for all below listed firmware variants. ip150.bin Protocol packets H.323 and PPP without SIP ip150-sip.bin Protocol packets SIP and PPP without H.323 ip150-no-ppp.bin Protocol packets H.323 and SIP without PPP The variant must be selected according to the current application. Telephones with these firmware variants can be loaded again with an older firmware and bootcode (V7, V8 or V9). Frequently asked Questions / Trouble Shooting Page 11

12 1.3.4 Bootcode and Firmware Update At the main version V6 update the bootcode or firmware if necessary. At the main version V7 or higher update the firmware and the bootcode. 1.4 Setup of the Language of the User Interface of the Telephone The setup of the language of the user interface of the telephone can be done in different ways Setup of the Language via DHCP Server If the telephone is working in DHCP client mode, then the language of the user interface of the telephone can be advised as option together with the IP address of the telephone by the DHCP server. A via DHCP server advised language supersedes the local setups of the telephone Setup of the Language at the Telephone The setup of the language is referred to a registration. The language setup will be carried out with: Menu->User List->Registration x (with x=1 6)->Preferences->Language With the cursor right and cursor left keys a language can be selected. After pressing the menu key twice a security question appears to store the changes. After an acknowledgement with yes the setup will be assumed, otherwise it will be dropped. At the telephone the language setup of the actual registration will appear. Therefore it has to be checked which one is the active registration. If more than one registration is configured and for all registrations the same language shall be available, then the language setup of each registration has to be set to the same language. Page 12 Frequently asked Questions / Trouble Shooting

13 1.4.3 Setup of the Language with the Web Interface The setup of the language is referred to a registration. The language setup will be carried out with: Software version V6, V7 or V8: Configuration->Registration x (with x=1 6)->Preferences->Language Software version V9 or higher: Phone->User-x (with x=1 6)->Preferences->Language The setup can be done with a combo box that has to be acknowledged with clicking at the OK button. At the telephone the language setup of the actual registration will appear. Therefore it has to be checked which one is the active registration. If more than one registration is configured and for all registrations the same language shall be available, then the language setup of each registration has to be set to the same language. The setup of the language will be carried out at the telephone with the selected language of the active registration. The setup of the language will be carried out at the web interface in English, because the language of the web interface can t be changed. This is always an advantage, when the user interface of the telephone is set to a language that cannot be understood. If the phone is working as DHCP client and the phone has no IP address, then it is impossible to change the language. Frequently asked Questions / Trouble Shooting Page 13

14 2 Troubleshooting 2.1 Commissioning 1. The device doesn t go on line after being connected. According to the device the power supply will be carried out via Power over Ethernet (PoE) or external supply. Check the cable connections and the power supply. If the device has a LAN connector at the housing, then check after dismounting the keypad / housing cover the connection cable between the printed board und the female LAN play at the housing being plugged in correctly. If the device has more than one LAN connector, then check the power supply being connected to the correct plug in. The details you can find in the manual of the respective device. According to the device the power via Power over Ethernet (PoE) will be accepted at the spare lines only or the used data and the spare lines. If you have a phone accepting the power via Power over Ethernet (PoE) at the spare lines only, be sure that your PoE injector is using the spare lines for power injection. With PoE there are different PoE classes defined. Be sure, that the PoE injector will accept the PoE Class of the device to be powered. A class 0 device needing about 3W, may be not powered by PoE injectors supporting class 1 and/or class 2 devices only, because a class 0 device is allowed to use maximal 12.96W. 2. The device cannot be operated with the keypad. After dismounting the keypad / housing cover check the connection cable between the printed board and the keypad being plugged in correctly. Page 14 Frequently asked Questions / Trouble Shooting

15 3. The device has no IP address. The device can be used with a fixed IP address, as DHCP client or as DHCP server. In the initial condition the device is working as DHCP client. Pay attention your DHCP server is working in a matter, so that it is possible to advise an IP address to the device. If this is not possible or you have no DHCP server, the device must go on line with a fixed IP address, IP mask and gateway address. You have to ask your network administrator. 4. After commissioning the phone the language of the user interface can t be changed. On default the device will be delivered in the DHCP client mode. At this the device can be sent important parameter for commissioning by the DHCP server (e. g. gatekeeper address, language of the user interface, etc.). While the device is working as DHCP client mode and has no IP address has been advised, the language of the user interface can t be changed at the phone or at the web interface. If the language of the user interface shall be changed before entering parameters, the DHCP client mode has to be disabled. Afterwards the language can be changed at the phone locally. If the language shall be changed via the web interface the phone has to be programmed first with an IP address, mask and, if necessary, a gateway address. After that the DHCP client mode can be activated again. 5. After commissioning without pressing a key the illumination of the display is flashing periodic. The device has no longer a program (memory error) and must be sent back for repairing to the manufacturer. 2.2 Registration 1. After going on line the phone has no successful registration. Check, that the device has an IP address, IP mask and a gateway address. Check, that a gatekeeper/pbx address is configured. Check, if one of the protocols SIP, TSIP or SIPS shall be used, that for registration another than the default port number is expected. Some gatekeeper/pbx expect that for registering of a standard SIP telephone another port is used than for registering of a SIP system telephone. Frequently asked Questions / Trouble Shooting Page 15

16 If the device is working as DHCP client, the DHCP server can advise the gatekeeper/pbx address to the phone. Check the advised parameter by the DHCP server. If the protocol is set to H.323 and no gatekeeper/pbx address is configured, a gatekeeper recovery is implemented. If you have more than one gatekeeper in the network, check that have a registration with the right gatekeeper. Check using the in the device implemented diagnostic functions ping and trace route the availability of the gatekeeper/pbx in the network. Check that you have a free licence at the gatekeeper/pbx for a registration of an additional device. If necessary you may need additional licenses for special protocols, protocol options or devices of third party manufacturers. You have to ask your network administrator. The registration parameter at the phone must be according to estimated parameter by the gatekeeper/pbx. You have to ask your network administrator. If you have entered all possible parameters for a registration, then you should try to get a registration with as few parameters as fit for the first step. You should try to get a registration without setting a password at the phone and at the gatekeeper/pbx. If you use the Protocol SIP (TSIP, SIPS), you should pay attention that a SIP account can be configured in different variants. For details have a look to chapter 3. If you use the protocol SIPS, then you must have a NTP server configured reachable for the device. In this case date and time will be displayed at the phone. If you use the protocol SIPS, then the gatekeeper/pbx and the phone must be configured to approve the certificates one another. Check the configuration of telephone and gatekeeper/pbx. Page 16 Frequently asked Questions / Trouble Shooting

17 2.2.1 Port Number, Protocols SIP, TSIP and SIPS The protocols SIP and TSIP are using for the registration the default port number 5060 and the protocol SIPS is using for the registration the default port number Concerning to the software version and release the default port number can be changed by setting the optional parameter /gk-port, by appending of ":<port number>" to the domain and suppressing the proxy/alternate proxy address or by appending of ":<port number>" to the proxy respectively alternate proxy address. If it is not sure which variant of the setup for the software version and release is necessary, then the result has to be checked with a protocol analyser. 2.3 Making Calls 1. In face with an active registration you can make no outgoing call. If you use the protocol H.323, Then you have to check the gatekeeper/pbx to support the protocol options faststart, H.245- tunneling und extended fast connect and to support these protocol options with devices of other manufacturers too. Check the gatekeeper/pbx to support single dialling. If this is not possible, then you should enter for example the value 2 for the parameter enblock dialling timeout. If the subscriber has been setup newly, then you have to check to advised rights and the existing routing information for the subscriber. You have to ask your gatekeeper/pbx administrator. Check the configuration of the codec of the gatekeeper/pbx and the phone. 2. In face with an active registration you can t get an incoming call. If the subscriber has been setup newly, then you have to check to advised rights and the existing routing information for the subscriber. You have to ask your gatekeeper/pbx administrator. Check the configuration of the codec of the gatekeeper/pbx and the phone. Frequently asked Questions / Trouble Shooting Page 17

18 3. You can make a call (incoming, outgoing) but you have only a one-way or no speech connection. If you use the protocol H.323, Then you have to check the gatekeeper/pbx to support the protocol options faststart, H.245- tunneling und extended fast connect and to support these protocol options with devices of other manufacturers too. If necessary you have to disable these options. Page 18 Frequently asked Questions / Trouble Shooting

19 3 SIP Accounts The general registration parameters of a SIP account are: Domain (optional) CGPN (extension number) User ID Proxy (IP address or DNS name of the PBX) (Must be always configured.) At least one of the parameters CGPN or User ID must be configured. Concerning to the configured parameters the registration will be executed for the following SIP account. If in the parameters domain or proxy a DNS name is configured, for creating a SIP account it will not be translated into an IP address via DNS server. Domain CGPN User ID SIP Account IP Address for the Registration Message X X X User ID@Domain IP Address (Proxy) X X CGPN@Domain IP Address (Proxy) X X User ID@Domain (*) IP Address (Proxy) X X User ID@Proxy IP Address (Proxy) X CGPN@Proxy IP Address (Proxy) X User ID@Proxy (*) IP Address (Proxy) (*) In these cases the telephone can be dialled with a SIP account, but not with a calling number. Example: Domain: company.com User ID: max.mustermann Frequently asked Questions / Trouble Shooting Page 19

20 CGPN: 1234 Proxy: pbx.company.com, IP address: Domain CGPN User ID SIP Account IP Address for the Registration Message X X X max.mustermann@company.com X X 1234@company.com X X max.mustermann@company.com X X max.mustermann@pbx.company.com X 1234@pbx.company.com X max.mustermann@pbx.company.com The registration parameters must be configured at the PBX and at the telephone identically. Additional an optional authorization can be used for securing the registration. Username Password If the authorization is activated at the PBX, then the parameters username and password must be configured at the PBX and at the telephone identically. Page 20 Frequently asked Questions / Trouble Shooting

21 4 Specifics of PBX s 4.1 3CX (Software PBX) For a correct inquiry call the protocol option SIP-hold (see manual) has to be set to the value 1 (sendonly). 4.2 Avaya Call Manager Block Dialing The PBX Cisco Call Manager Express supports block dialing only. It is recommended to set the parameter Enblock Dialing Timeout via web interface to the value 2 or 3. The parameter can be found depending on the software version on the following page. SW V6, V7, V8: /Registration x/registration with x:1 6 SW V9 or higher: Phone/User-x/General with x: Licensing At the Avaya Call Manager separate from the normal licensing of the PBX you need for each connected device of a third party manufacturer an additional special license Protocols H.323 The Avaya Call Manager with an older Software Version supports the protokoll H.323 only. With newer software versions the protocol SIP will be supported too. The correct operation can be garanteed only, if the Avaya PBX has a software release July 2010 or later SIP With newer software releases the protocol SIP will be supported too. The Avaya PBX should have a software release April 2012 or later. Frequently asked Questions / Trouble Shooting Page 21

22 4.3 Cisco Call Manager Express Block Dialing The PBX Cisco Call Manager Express supports block dialing only. It is recommended to set the parameter Enblock Dialing Timeout via web interface to the value 2 or 3. The parameter can be found depending on the software version on the following page. SW V6, V7, V8: /Registration x/registration with x:1 6 SW V9 or higher: Phone/User-x with x: Unified Communications Manager (Call Manager) Block Dialing The PBX Cisco Unified Communications Manager supports block dialing only. It is recommended to set the parameter Enblock Dialing Timeout via web interface to the value 2 or 3. The parameter can be found depending on the software version on the following page: SW V6, V7, V8: /Registration x/registration with x:1 6 SW V9 or higher: Phone/User-x with x: Configuring Unified Communications Manager for Non-Cisco SIP Phones Cisco Unified Communications Manager (CUCM) supports Cisco SIP IP Phones as well as PFC3261 compliant SIP phones from third party companies. This chapter describes some information how to configure third party SIP phones by using Cisco Unified Communications Manager Administration. Page 22 Frequently asked Questions / Trouble Shooting

23 Feature Integrated with Centralized TFTP Sends MAC Address Downloads Softkey File Downloads Dial Plan File Supports Cisco Unified Communications Manager Failover and Fallback Supports Reset and Restart Table 1: Non-Cisco SIP Phone Configuration Setting No No No No No No When third-party SIP phones get configured, the Cisco Unified Communications Manager database gets updated when the administrator uses Cisco Unified Communications Manager Administration. The administrator must also perform configuration steps on the third-party SIP phone; see following examples: Proxy address in the phone should be the IP or Fully Qualified Domain Name (FQDN) of the Cisco Unified Communications Manager. Directory number(s) in the phone should match the directory number(s) that are configured for the device in Cisco Unified Communications Manager Administration. Digest user ID (sometimes referred to as Authorization ID) in the phone should match the Digest User ID in Cisco Unified Communications Manager Administration. See the "Third-Party SIP Phone Configuration Checklist" for the Cisco Unified Communications Manager Administration configuration steps How Cisco Unified Communications Manager identifies a Third-Party SIP Phone Because third-party SIP phones do not send a MAC address (as layer 7 information), they must identify themselves by using digest authentication. The REGISTER message includes the following header: Authorization: Digest username="swhite",realm="ccmsipline",nonce="gbauadss2qowr6k9y3hggv DAqnLfoLk5",uri="sip: ",algorithm=MD5,response="126c0643a ab59d4f e" Frequently asked Questions / Trouble Shooting Page 23

24 The username, swhite, must match an end user that is configured in the End User Configuration window of Cisco Unified Communications Manager Administration (see Adding an End User in the Cisco Unified Communications Manager Administration Guide). The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window (see Configuring Cisco IP Phones in the Cisco Unified Communications Manager Administration Guide). You can assign each end user ID to only one third-party phone (in the Digest User field of the Phone Configuration window). If the same end user ID is assigned as the Digest User for multiple phones, the third-party phones to which they are assigned will not successfully register Third-Party SIP Phones and TFTP Third-party SIP phones do not get configured by using the Cisco Unified Communications Manager TFTP server. The customer configures them by using the native phone configuration mechanism (usually a web page or a configuration file). The customer must keep the device and line configuration in the Cisco Unified Communications Manager database synchronized with the native phone configuration (for example, extension 1002 on the phone and 1002 in Cisco Unified Communications Manager). Additionally, if the directory number of a line is changed, ensure that it is changed in both Cisco Unified Communications Manager Administration and in the native phone configuration mechanism Enabling Digest Authentication for Third-Party SIP Phones To enable digest authentication for third-party SIP phones, the administrator must create a SIP Phone Security Profile. See the Cisco Unified Communications Manager Security Guide for details. On the SIP Phone Security Profile Configuration window, check the Enable Digest Authentication check box. After the security profile is configured, the administrator must assign that security profile to the SIP phone by using the Phone Configuration window. If this check box is not checked, Cisco Unified Communications Manager will use digest authentication for purposes of identifying the phone by the end user ID, and it will not verify the digest password. If the check box is checked, Cisco Unified Communications Manager will verify the password. Page 24 Frequently asked Questions / Trouble Shooting

25 Cisco Unified Communications Manager does not support Transport Layer Security (TLS) from third-party SIP phones Third-Party SIP Phone Configuration Checklist The following table provides steps to manually configure a third-party SIP phone by using Cisco Unified Communications Manager Administration. Procedure Step 1 Step 2 Step 3 Step 4 Gather the following Information about the phone: MAC address Physical location of the phone Cisco Unified Communications Manager user to associate with the phone Partition, calling search space, and location information, if used Number of lines and associated DNs to assign to the phone Determine whether sufficient Device License Units are available. If not, purchase and install additional Device License Units. Third-Party SIP Devices (Basic) and (Advanced) consume three and six Device License Units each, respectively. See topics related to calculating the number of required licenses and obtaining a license in the Cisco Unified Communications Manager Features and Services Guide. Configure the end user that will be the Digest User. Note: If the third-party SIP phone does not support an authorization ID (digest user), create a user with a user ID that matches the DN of the third-party phone. For example, create an end user named 1000 and create a DN of 1000 for the phone. Assign this user to the phone (see Step 9). Configure the SIP Profile or use the default profile. The SIP Profile gets added to the SIP phone by using the Phone Configuration window. Note: Third-party SIP phones use only the SIP Profile Information section of the SIP Profile Configuration window. Frequently asked Questions / Trouble Shooting Page 25

26 Procedure Step 5 Step 6 Step 7 Step 8 Step 9 Step 10 Configure the Phone Security Profile. To use digest authentication, you must configure a new phone security profile. If you use one of the standard non-secure SIP profiles that are provided for auto-registration, you cannot enable digest authentication. Add and configure the third-party SIP phone by choosing Third-party SIP Device (Advanced) or (Basic) from the Add a New Phone Configuration window. Note: Third-party SIP Device (Basic) supports one line and consumes three license units, and Third-party SIP Device (Advanced) supports up to eight lines and video, and consumes six license units. Add and configure lines (DNs) on the phone. In the End User Configuration window, associate the third-party SIP phone with the user by using Device Association and choosing the SIP phone. In the Digest User field of the Phone Configuration window, choose the end user that you created in Step 3 Provide power, install, verify network connectivity, and configure network settings for the third-party SIP phone. Step 11 Make calls with the third-party SIP phone. See the user guide that came with your third-party SIP phone. Table 2: Third-Party SIP Phone Configuration Checklist DTMF Reception To require DTMF reception, check the Require DTMF Reception check box that displays on the Phone Configuration window in Cisco Unified Communications Manager Administration. Page 26 Frequently asked Questions / Trouble Shooting

27 Licensing Third-Party SIP Phones Licensing of third-party phones that are running SIP enforces the following limitations: Third-party SIP Device (Basic) Video calls do not get supported. Video enforcement occurs as part of the offer/answer process. If video-related media is provided as part of an offer or answer from a SIP device that is not permitted to negotiate video, only the non-video-related parts of the call get extended to the destination party. Similarly, a SIP endpoint that is not permitted to negotiate media will not receive any video-related media in the SDP that is sent from Cisco Unified Communications Manager. Third-party SIP Device (Advanced) and (Basic) Cisco-specific SIP extensions do not get supported. Some Cisco-specific SIP extensions that are not supported include service URIs, header extensions, dialog subscriptions, and remote call control proprietary mime types. Cisco Unified Communications Manager will reject any request from a phone that is running SIP that is not permitted to use an advanced feature that uses a service request URI (such as Call Pickup URI, Meet Me Service URI). The SIP profile specifies service URIs. The profile gets assigned to SIP devices. Cisco Unified Communications Manager will block features that require the use of Cisco-specific SIP extensions. Ensure that any wireless third-party SIP client or device is configured as a Third-Party SIP Device (Advanced) in conformance with Cisco Unified Communications Manager licensing policy. In Cisco Unified Communications Manager, Release 5.1(1) and above, certain characteristics for Basic and Advanced Third-Party phones that are running SIP changed. These characteristics include changes to the Maximum Number of Calls per Device, Default Maximum number of calls per DN, and Default Busy Trigger per DN fields that display on the Directory Number Configuration window in Cisco Unified Communications Manager Administration. The following tables provide more information. Frequently asked Questions / Trouble Shooting Page 27

28 Field Name Old Value New Value Maximum Number of Calls Per Device Default Maximum Number of Calls per DN Default Busy Trigger per DN 2 2 Table 3: Directory number migration changes for basic third-party phones that are running SIP Field Name Old Value New Value Maximum Number of Calls Per Device Default Maximum Number of Calls per DN Default Busy Trigger per DN 2 2 Table 4: Directory number migration changes for advanced third-party phones that are running SIP For users that have third-party phones that are running SIP that are configured on any version of release 5.0 that are migrating/upgrading to release 6.0(1) or above, be aware that, after the upgrade, these devices retain their release 5.0 configured values. However, if users need to make changes to DN configuration values, users must change Maximum Number of Calls and Default Busy Trigger values on each DN. For basic third-party phones that are running SIP, only one line value needs to be modified. However, for advanced third-party phones that are running SIP, users potentially must disassociate lines on the device before they can make any DNrelated configuration changes. This situation potentially can happen if more than four lines are configured. An example scenario follows: Advanced phone configured with 6 lines with Maximum number of calls = 4 and Busy Trigger = 2 for each line. Page 28 Frequently asked Questions / Trouble Shooting

29 After upgrade to release 6.1, ensure maximum number of calls on the device is reduced to 16 or below before any DN changes. The current value on this phone equals 24 (6 lines * 4). The device essentially exists in a negative zone (16-24). User would disassociate two lines from the device. After the user disassociates those lines from the device, you can modify the DN characteristics for the remaining four lines by setting Maximum Number of Calls and Busy Trigger to an appropriate value. User re-associates the disassociated lines Where to find more Information Directory Number Configuration, Cisco Unified Communications Manager Administration Guide Cisco IP Phone Configuration, Cisco Unified Communications Manager Administration Guide SIP Profile Configuration, Cisco Unified Communications Manager Administration Guide End User Configuration, Cisco Unified Communications Manager Administration Guide Cisco IP Phones, Cisco Unified Communications Manager System Guide The documentation can be found on the homepage of Cisco. 4.4 NEC SV General At a NEC SV8100 PBX H.323 telephones, SIP telephones and NEC SIP system telephones can be registered. NEC SIP system telephones are SIP telephones with private SIP protocol extensions. Frequently asked Questions / Trouble Shooting Page 29

30 Characteristics With the first registration of a VoIP telephone the type of the telephone will be stored inside the PBX. If another telephone should be registered for the same extension, a telephone with the same type has to be used or the stored type of a telephone in the PBX must be deleted. If no Type is stored or the type is MEGACO any telephone can be registered Port Numbers for the Registration Each type of end device must be registered at a different port at the PBX. For the NEC SV8100 PBX there are the following default settings, which can be changed if necessary in the PBX. type of end device port number H.323 line 1720 H.323 terminal 1720 SIP line 5060 SIP terminal 5070 SIP system terminal Supplementary Services and Limitations With H.323 connections only simple call connections can be used. There are only a few supplementary services (e. g. calling number presentation) available. With a H.323 terminal no inquiry call, no call transfer, etc. is possible. With SIP connections a lot of supplementary services can be used. To use all supplementary services SIP system connections have to be used. Page 30 Frequently asked Questions / Trouble Shooting

31 To an end device it is only supported to use one media stream. Therefore it is not possible to use a local conference at a phone. Telephone conferences can only be used with the supplementary service conference in the PBX. A lot of supplementary services (e. g.: conference, call forwarding, etc.) will be controlled with feature codes. Details can be seen in the documentation of the NEC SV8100 PBX Deleting the Telephone Type in the PBX The deleting the phone type of an extension in the PBX can t be done with the web interface of the PBX. It can be executed with a system phone as described below: Press the loudspeaker key Enter #*#* Enter the PBX password from the web interface of the PBX Press the Transfer key Enter 9023 Press the softkey Select Select the phone you want to delete the type with the + / - softkeys Press the Mic key Enter 1 Press the Transfer key Press the Mic key Press the Loudspeaker key Setup for FHF Telephones (FernTel IP, (Ex)ResistTel IP2) General The parameter Enblock Dialing Time Out must be set (value: 2 4). Frequently asked Questions / Trouble Shooting Page 31

32 Protocol H.323 The protocol options /no-faststart, /no-h245-tuneling and /no-efc are to be set if necessary Protocol SIP At the registration parameters at proxy <IP address of the PBX>:<port number> e. g.: :5070 has to be set Software Version V8 To use the SIP protocol with a FHF telephone, the FHF telephone with a version V8 software must use a release [80818] or higher Software Version V9 To use the SIP protocol with a FHF telephone, the FHF telephone with a version V8 software must use a release [ ] or higher. Additional the protocol option "/c-line-at-session-level" must be activated Software Version V10 or higher To use the SIP protocol with a FHF telephone, at the FHF with the software version V10 or higher the protocol option "/c-line-at-session-level" must be activated Changing the Protocol After changing the protocol from H.323 to SIP or reverse the type of telephone must be cleared in the PBX. Then the FHF telephone has to be restarted again Starting a Conference from a FHF Telephone The conference must be controlled in the NEC PBX. The default feature code for conference is 826. The FHF telephone is calling another Telephone or is called by another telephone. At the FHF telephone the R key has to be pressed and with 826nnn (with nnn = number of another telephone) an additional telephone will be called. By pressing the R key four times the conference will be started and the called telephone will be taken into the conference. With the same manner additional telephones can be taken into the conference. Page 32 Frequently asked Questions / Trouble Shooting

33 The local conference at the FernTel IP/(Ex)ResistTel IP2 has to be locked. Therefore by using the web interface of the phone at the page Phone/Protect in the field Fine grained Function Locking the value 0x (with an OR composition to maybe other values) has to be added. For details have a look to the manual of the FernTel IP/(Ex)ResistTel IP Call Forwarding The call forwarding must be controlled in the NEC PBX. The default feature code for call forwarding is 848. By dialling of 8481nnn (with nnn = number of the selected telephone) the call forwarding will be selected. When in the display of the FHF telephone the text Getting Channel appears the phone can be hanged up again. The call forwarding is now active. With dialling of 8480 the call forwarding can be cleared. With using instead of 848 another feature code of the PBX then another variant of the call forwarding can be selected. The special variant of a local call forwarding in the FernTel IP/(Ex)ResistTel IP2 with the protocol SIP must not be enabled in the configuration. Frequently asked Questions / Trouble Shooting Page 33

34 5 Power over Ethernet (PoE) 5.1 FernTel IP Family All telephones of the FernTel IP family have the PoE signature of a class 0 device. Although the device needs maximal 3.5 W power there may be problems with PoE injectors supporting only class 1 (3.84 W) or class 2 (6.49 W) devices. These PoE injectors recognise the danger a class 0 device allowed to consume maximal W can overload the PoE injector. Therefore at the PoE injector the PoE voltage will not be switched on for a device that maybe can overload it. 5.2 ResistTel IP2 Family All telephones of the ResistTel IP2 family have the PoE signature of a class 0 device and they need maximal W. Some PoE injectors supporting some PoE ports support class 0 devices, but don t have enough power for supporting devices with the maximal power at all ports. At injectors with power management it is possible, that a new additional device will not be commissioned, because no further power is available. At injectors without power management it is possible, that will be overloaded. This may result a breakdown of the PoE supply. 5.3 ExResistTel IP2 (INMETRO, EAC) Family All telephones of the ExResistTel IP2 (INMETRO, EAC) family have the PoE signature of a class 0 device and they need maximal W. To observe the ATEX directives for zone 1 devices, it is permissible to transmit the PoE voltage with the unused data lines only. E. g. the following PoE injectors respectively switches with integrated PoE injectors are usable for operating of an ExResistTel IP2 (INMETRO, EAC) telephone. 1 channel PoE injector, manufacturer: Intellinet Network Solution, model No channel PoE injector, manufacturer: Planet Networking & Communication, model: POE-1200G 24 channel PoE injector, manufacturer: Planet Networking & Communication, model: POE-2400G 24 port PoE switch, manufacturer: Planet Networking & Communication, model FNSW-2400PS Page 34 Frequently asked Questions / Trouble Shooting

35 1 channel PoE injector, manufacturer: Red Lion Controls, Modell: EB-PSE-24V- 1A 2 channel PoE injector, manufacturer: Red Lion Controls, Modell: EB-PSE-48V- 2A 2 channel PoE switch, manufacturer: Ethernet Direct, Modell Cobra CUE-111, 1 Port 10/100 Base RJ-45, 1 Fibre Port 100 Base Multi Mode (SC Connector) 2 channel PoE switch, manufacturer: Ethernet Direct, Modell Cobra CUE-113, 1 Port 10/100 Base RJ-45, 1 Fibre Port 100 Base Single Mode (SC Connector) 1 channel PoE injector, manufacturer: MTL-Instruments (Eaton), Modell: 9299-PSE24A 2 channel PoE injector, manufacturer: MTL-Instruments (Eaton), Modell: 9299-PSE48A 16/18 channel PoE switch, manufacturer: Cisco, Modell: Industrial Ethernet 2000 Series Switches o IE PTC-G-L o IE PTC-G-E o IE PTC-G-NX Frequently asked Questions / Trouble Shooting Page 35

36 6 Ethernet Connection over a greater Distance 6.1 Ethernet Extender The maximal length of a (PoE) copper Ethernet cable from the switch to the telephone is 100m. A considerably increase of this distance provide Ethernet (PoE) extender. On the market there are devices available supporting PoE devices with remote current via 2 wire twisted pair or coaxial cable over a distance up to 3000m. An example is the fast Ethernet exp-s110-xt PoE extender of the company Perle Systems Limited. This device is a 10/100 Mbit/s PoE copper extender for industrial temperatures (-40 C C). 6.2 WLAN With a WLAN client Ethernet adapter a VoIP telephone can be connected into a WLAN infrastructure. The WLAN client Ethernet adapter and the telephone then must be powered external only. With an applicable selection of the antenna, an inter visibility between the antenna and respecting the maximal lawful allowed electric radiated power (country specific directives) a distance up to 35km can be bridged. 6.3 Optical Ethernet (Fibre Optic Cable) For bridging of very great distances a conversion with a fibre optic converter is provided. With fibre optics cables (FOC) a distance up to 100km can be bridged without additional amplifying. Page 36 Frequently asked Questions / Trouble Shooting

37 7 External Signalling Devices 7.1 Connection of the weatherproof Signalling Device TWIN- Kombi / Type 5843/5 to weatherproof VoIP Telephones The connection is only possible, if the VoIP telephone is equipped with a relay (closing) with potential free contacts Setup of the Ringing Volume Figure 1: Wiring Diagram of the Signalling Device TWIN-Kombi / FSK Type 5843/5 The ringing sound pressure level can be individually adjusted by means of the internal volume control (see Figure 1) Setup of the Operating Mode Use the jumpers (see arrow) to set the operating mode of the TWIN-Kombi. Frequently asked Questions / Trouble Shooting Page 37

38 Figure 2: Operation Mode analogue Telephone of the TWIN-Kombi / FSK Type 5843/5 Figure 3: Operation Mode Device with dry Contact of the TWIN-Kombi / FSK Type 5843/5 The operation mode for using the signalling device with a VoIP telephone must be configured to device with dry contact. By operating the TWIN-Kombi with 230V / 50 Hz there are at the terminals W and L b about 19,5 V AC. The short circuit current with closed potential free contact is about 3.15 ma AC. Page 38 Frequently asked Questions / Trouble Shooting

39 7.1.3 Setup of the Ringing Signal Use the DIP switch to set the ringing signal of the TWIN-Kombi. Switch position Tone sequence frequency [khz] Tone sequence Nr. 1 Nr. 2 1,4 1,5 2,2 2,9 [sec] 0 0 X X 0,3 0 1 X X 0,3 1 0 X X 0,5 1 1 X X X 0, Setup of the Telephone FernTel IP Family At a telephone FernTel IP with relay muss the telephone must be configured to a blinking keypad for an incoming call ResistTel IP2 Family At a telephone ResistTel IP2 with relay module the relay must be configured to 'Alert on' ExResistTel IP2 (INMETRO, EAC) Family At a telephone ExResistTel IP2 (INMETRO, EAC) the relay must be configured to 'Alert on'. The operating of the explosion proof telephone is not allowed in hazardous areas. 7.2 Connection of the explosion proof TWIN-EExII Type 5842/1 to a VoIP Telephone The connection is only possible, if the VoIP telephone is equipped with a relay (closing) with potential free contacts. Frequently asked Questions / Trouble Shooting Page 39

40 7.2.1 Setup of the DIP switch (Operating Mode / Melody) Figure 4: DIP Switch Operating Mode / Melody of the TWIN-EExII Type 5842/1 The operating mode for using the signalling device with a VoIP telephone must be configured to signal unit. Page 40 Frequently asked Questions / Trouble Shooting

41 7.2.2 Terminal Connections Figure 5: Terminal Connections of the TWIN-EExII Type 5842/1 The terminals W and L b are for using the signalling device as signal unit without relevance Wiring With the potential free contact of the relay in the telephone the power supply of the signalling device will be switched on or off. For the necessary wiring of the feeder, the relay contacts and the connection for power supply of the signalling device a junction box has to be used. Frequently asked Questions / Trouble Shooting Page 41

42 7.2.4 Setup of the Telephone ExResistTel IP2 (INMETRO, EAC) Family At a ExResistTel IP2 (INMETRO, EAC) the relay must be configured to 'Alert on'. Page 42 Frequently asked Questions / Trouble Shooting

43 8 Device internal Troubleshooting For searching errors you can use the implemented diagnostic possibilities inside the device. Ping Traceroute Logging Tracing Reading out of events and alarms Reading out of the configuration of the device Reading out of the via DHCP configured parameter The methods logging and tracing can change the timing of the phone, the fault scenario can change and an error diagnostic will be no longer possible. 8.1 Tracing By using the tool trace(contiguous) there are sometimes problem that the display of the information on the screen appears with a large delay (up to 3 minutes). The case is an installed and working virus scanner with an integrated browser protection. In this case the IP address of the monitored telephone must be configured as an exception for the browser protection. Frequently asked Questions / Trouble Shooting Page 43

44 9 Wireshark 9.1 General Another possibility to determine protocol problems is the program Wireshark. This open source program is a protocol analyser with an English user interface. It can be downloaded from the internet at and used without any cost. The program is available for different operation systems e. g. Windows or Linux. To analyse the complete message traffic from and to the phone you have to use the following setup. IP telephone PoE injector Hub LAN PC Wireshark Gatekeeper / PBX IP telephone 2 IP telephone 3 Figure 1: Setup for Wireshark Trace It is important to connect the PC with the program Wireshark with a hub with LAN connection of the phone. An unmanaged switch is inapplicable. The experience with a managed switch showed, that often at the mirror channel not all messages can be seen and therefore an analysis of the messages is impossible. 9.2 RPCAP Server and Microsoft Windows A special possibility is to use the phone as RPCAP Server and to capture messages with the program Wireshark at the phone, without connecting the phone to the LAN as described in Figure 1. This possibility is only available for Windows PCs. Page 44 Frequently asked Questions / Trouble Shooting

45 At the Windows PC the program Wireshark must be installed. Additional the file innovaphone.dll must be copied into the Wireshark plugins directory. (e. g.: c:\programme\wireshark\plugins\1.2.5). Pay attention to the actual installed version of the program Wireshark and use only the file innovaphone.dll according to your operating system (32-bit Windows, 64-bit Windows) Configuration of the RPCAP Server The remote PCAP server is disabled on default. To enable it the flag Enable RPCAP must be activated at the web interface of the phone at Administration->Diagnostics- >Tracing. To capture all IP traffic the flag All TCP/UDP Traffic has to be activated. Otherwise only the trace flags of to be captured modules have to be activated Capturing with the Program Wireshark Start the program Wireshark and open the capture options dialog Older Wireshark Versions Enter "rpcap://<ip address of the phone>/trace" into the interface field. Then click the button start to start the capturing Newer Wireshark Versions Inside of the Capture Options dialog open the Manage Interfaces Dialog. Inside the register card Pipes enter a new pipe with Enter "rpcap://<ip address of the phone>/trace". After closing the window a new entry appears in the interface list. Activate this item. Then click the button start to start the capturing. Frequently asked Questions / Trouble Shooting Page 45

46 10 Utility Programs 10.1 SoftPerfect Network Scanner This program is a freeware Windows network scanner with an English user interface. All active devices in a network segment responding to an ICMP protocol request can be recognized. It can be downloaded from the internet at and used without any cost. If the PC is connected to the network segment to be checked, then for each device it will be displayed the MAC address, the IP address and if applicable other parameters. The program is very helpful for remote supporting of a device with known MAC address to recognize the IP address distributed by a DHCP server without getting the IP address from the device locally or with an administrative access to the DHCP server Tera Term Professional This program is a freeware Windows terminal emulation with an English user interface. It will be supported Telnet and SSH1 respectively SSH2 connections. It can be downloaded from the internet at and used without any cost TFTPD32 This is a European Public License Windows program with an English user interface to be used as a TFTP, SNTP, Syslog, DHCP and DNS Server respectively as TFTP client. It can be downloaded from the internet at and used without any cost. The program supports server and client functionalities being not available with the operating system Windows exceptionally the server versions Known Problems Service cannot be started If an error message appears during starting of the program, that e. g. the DHCP server service can t be bonded to a special port, then in most cases the reason is, that another service is active at this port. Then the service using this port must be stopped. Page 46 Frequently asked Questions / Trouble Shooting

47 Problems storing the Configuration With Windows 7 there are sometimes problems storing the configuration. The reason is that the program was written for the operating system Windows XP. The configuration will be stored in a *.ini file located in the installation directory of the program (mostly C:\program files\...) With Windows 7 these directories are write protected. With Windows 7 the write protection for the *.ini file must be cleared or the program must be installed in a storage location Windows 7 is not setting the write protection automatically OpenLDAP This program is a free, open source implementation of the Lightweight Directory Access Protocol (LDAP) with an English user interface. The program is also available for the operating system Windows. The Windows version can be downloaded from the internet at and used without any cost. This program can be used for realizing an external directory on a Windows PC for devices with a LDAP interface Linphone This program is a GNU Public License SIP Video Softphone with an English user interface. The program is also available for the operating system Windows. It can be downloaded from the internet at and used without any cost. This program is helpful for testing, if an additional simple SIP telephone will be needed H323Phone This program is a Freeware H.323 Windows Softphone with an English user interface. It can be downloaded from the internet at and used without any cost. This program is helpful for testing, if an additional simple H.323 telephone will be needed WirelessNetView This program is a freeware Windows program with an English user interface. It can be downloaded from the internet at and used without any cost. This program can be used to check the available WLAN s at a place. This program is helpful for commissioning a VoIP telephone connected to an external WLAN access point. Frequently asked Questions / Trouble Shooting Page 47

48 10.8 Additional IP Address at an Ethernet Interface Sometimes a VoIP telephone must be configured, that has a fixed IP address not located in the local network area of a service laptop. In these cases it will be helpful to setup an additional fixed IP address and an IP address mask at the service laptop. This can be done by using the net shell commands in a command window or by using the extended TCP/IP setup of the internet properties in the system control. Such a configuration saves the permanent changing of the network configuration. Page 48 Frequently asked Questions / Trouble Shooting

49 11 Abbreviations X Standard for Authorisation in Computer Networks (IEEE) 802.3af Standard for Power over Ethernet (IEEE) A A AAA AAAA AC ACL ACM A/D AD ADC AES ANSI AOR API ARP AS-SIP ASCII ASN.1 ATEX ATM B BER C CA CAPI CAS CBIPS CCP CDPN CDR CE CEC CED CER DNS-RR Address Record Authentication, Authorization and Accounting DNS-RR IPv6 Address Record Alternate Current Access Control List Avaya Call Manager Analogue/Digital Active Directory Analogue Digital Converter Advanced Encryption Standard American National Standards Institute Account of Registration Application Program Interface Address Resolution Protocol Assured Services-Session Initiation Protocol American Standard Code for Information Interchange Abstract Syntax Notation One Atmosphere Explosive Asynchronous Transfer Mode Basic Encoding Rules (ASN.1) Certificate Authority Common ISDN Programming Interface Central Authentication Protocol Content-based IPS Compression Control Protocol Called Party Number Call Detail Records Conformité Européenne Canadian Electric Code Called Terminal Identification (FAX) Canonical Encoding Rules (ASN.1) Frequently asked Questions / Trouble Shooting Page 49

50 CEST CET CFR CGN CGPN CHAP CLIR CM CME CNAME CNG CNG CPN CQST CRC CRL CSA CSS CSTA CTI CUCM CUTR D DAC DC DCN DCS DDoS DECT DER DER DES DHCP DIN DIS DOM DoS DN DNS DNSBL DNS-RR Central European Summer Time Central European Time Confirmation to Receive Calling Party Number Calling Party Number Challenge Handshake Authentication Protocol Calling Line Identification Restriction Call Manager, siehe auch CUCM Call Manager Express DNS-RR Canonical Name Record Comfort Noise Generation Calling Tone (FAX) Calling Party Number China National Quality Supervision and Test Centre for Explosion Protected Electrical Products Cyclic Redundancy Check Certificate Revocation List Canadian Standards Association Cascading Style Sheets Computer Supported Telecommunications Applications Computer Telephony Integration Cisco Unified Communications Manager, siehe auch CM Customs Union Technical Regulations Digital Analogue Converter Direct Current Disconnect (FAX) Digital Command Signal (FAX) Distributed Denial of Service Digital Enhanced Cordless Telecommunications File name extension for a base64 coded certificate Distinguished Encoding Rules (ASN.1) Data Encryption Standard Dynamic Host Configuration Protocol Deutsches Institut für Normung Digital Identification Signal (FAX) Document Object Model Denial of Service Directory Number Domain Name Server DNS-based Blackhole List, Block List, or Blacklist DNS Resource Record Page 50 Frequently asked Questions / Trouble Shooting

51 DNV Det Norske Veritas DRAM Dynamic Random Access Memory DSS1 Digital Subscriber System No. 1 DSL Digital Subscriber Line DTLS Datagram Transport Layer Security DUID DHCP Unique Identifier E E.164 Standard for the international public telecommunication numbering plan (ITU-T) EAC Eurasian Conformity EAP Extensible Authentication Protocol EAR Foundation Elektro-Altgeraete Register ECN Encoding Control Notation (ASN.1) EDSS1 European Digital Subscriber System No. 1 EIA Electronic Industries Alliance EMC Electromagnetic Compatibility EMV Elektromagnetische Verträglichkeit ENUM Telephone Number Mapping EPID Endpoint Identifier ETSI European Telecommunication Standards Institute EUI bit Extended Unique Identifier F F/FTP FAT FAT32 FAX FCC FOC FoIP FQDN FQTN FTP FTP G GOST GMT GRE GSER GSM Foiled / Foiled Twisted Pair Cable File Allocation Table. A from Microsoft developed file system. A FAT Variant Telefax, Fernkopie Federal Communications Commission Fibre Optic Cable FAX over IP Fully Qualified Domain Name Fully Qualified Telephone Number File Transfer Protocol Foiled Twisted Pair Cable Gosudarstwenny Standart, sowjetische bzw. russische Normen Greenwich Mean Time Generic Routing Encapsulation Generic String Encoding Rules (ASN.1) Global System for Mobile Communications Frequently asked Questions / Trouble Shooting Page 51

52 H H323 Abbreviation for H.323 H.323 Higher recommendation of the ITU for protocols HIP Host Identity Protocol HIPS Host-based IPS HTML Hypertext Markup Language HTTP Hypertext Transfer Protocol HTTPS Hypertext Transfer Protocol Secure I IAID Interface Association Identifier IANA Internet Assigned Numbers Authority ICANN Internet Corporation for Assigned Names and Numbers ICE Interactive Connectivity Establishment IDN Internationalized Domain Name IE Internet Explorer IEC International Electrotechnical Commission IECEx IEC System for Certification to Standards relating to equipment for use in Explosive Atmospheres IEEE Institute of Electrical and Electronics Engineers IEEE 802.1X Standard for Authorisation in Computer Networks IEEE 802.3af Standard for PoE IETF Internet Engineering Task Force IIS Internet Information Server IMS Integrated Management System INCITS International Comittee for Information Technology Standards INMETRO Instituto Nacional de Metrologia, Qualidade e Tecnologia (National Institute of Metrology, Quality and Technology) IP Internet Protocol IPS Intrusion Prevention System IPv4 Internet Protocol Version 4 IPv6 Internet Protocol Version 6 ISA International Society of Automation ISC Internet Systems Consortium ISDN Integrated Services Digital Network ISO International Standardization Organisation ISP Internet Service Provider ISRAM Internal Static Random Access Memory ITU International Telecommunication Union IVR Interactive Voice Response Page 52 Frequently asked Questions / Trouble Shooting

53 J JAVA JMS K Kerberos KDC KPML L LabVIEW LAN LCD LCP LDAP LLDP-DUs LLDP-MED LID LLDP LTE LWL Java is a object orientated programming language Java Message Service Kerberos is a computer network authentication protocol Key Distribution Centre Keypad Markup Language Laboratory Virtual Instrumentation Engineering Workbench Local Area Network Liquid Crystal Display Link Control Protocol Lightweight Directory Access Protocol LLDP Data Units LLDP Media Endpoint Devices Language ID Link Layer Discovery Protocol Long Term Evolution Lichtwellenleiter (fibre optic cable) M MAC Message Authentication Code MAC-Address Media Access Control Address MCF Message Confirmation (FAX) MD5 Message Digest Algorithm 5 MEST Middle European Summer Time MET Middle European Time MIB Management Information Base MoH Music on Hold MOS Mean Opinion Score MPPC Microsoft Point-To-Point Compression Protocol MPPE Microsoft Point-To-Point Encryption Protocol MS Microsoft ms Millisecond MTBF Mean Time between Failures MTTF Mean Time to Failure MTTR Mean Time to Recover, Mean Time to Repair MTU Maximum Transmission Unit MX DNS-RR Mail Exchange Record Frequently asked Questions / Trouble Shooting Page 53

54 N NAC NAPTR NAS NAT NBNS NDP NEC NEMA NIPS NMS NS NTP O OBJ OID OID-IRI OUI-24 OUI-36 Network Access Control DNS-RR Naming Authority Pointer Network Access Server Network Address Translation NetBIOS Name Service Neighbour Discovery Protocol National Electrical Code National Electrical Manufacturers Association Network-based IPS Network Management System DNS-RR Name Server Record Network Time Protocol Object Identifier Object Identifier OID Internationalized Resource Identifier 24-bit Organizationally Unique Identifier 36-bit Organizationally Unique Identifier P P12 PKCS#12 file (see PFX) PA Potentialausgleich PAI P-Asserted Identity PAP Password Authentication Protocol PBX Private Branch Exchange PC Personal Computer PCAP PCAP (Packet Capture) free programming interface (API), to capture network traffic. PCM Pulse Code Modulation PDU Protocol Data Unit PEAP Protected Extensible Authentication Protocol PEM File name extension for a base64 coded certificate, enclosed with BEGIN CERTIFICATE and END CERTIFICATE PEN Private Enterprise Number PER Packed Encoding Rules (ASN.1) PFX Personal Information Exchange File (see P12) PHP5 Script Language PIN Personal Identification Number PKCS Private Key Cryptography Standards PKI Public Key Infrastructure PoE Power over Ethernet Page 54 Frequently asked Questions / Trouble Shooting

55 POSIX PPE PPI PPP PPPoE PPTP Proxy PSTN PTB PTR Q QoS QSIG R R&TTE RACS RADIUS RAS RBIPS RC4 RFC RID RLR RoHS RPCAP RSA RSIP RSSI RSTP RSVP RTCP RTP RTP/AVP RTP/AVPF RTP/SAVP RTP/SAVPF Portable Operating System Interface for UniX Personal Protective Equipment P-Preferred Identity Point-to-Point Protocol PPP over Ethernet Point-to-Point Tunnelling Protocol A hardware server that acts as an intermediary between a work station user and the Internet so that an enterprise can ensure security. Public Switched Telephone Network Physikalisch Technische Bundesanstalt DNS-RR Pointer Record Quality of Service Q-Interface Signalling Protocol Radio and Telecommunications Terminal Equipment Resource Access Control Facility Remote Authentication Dial-In User Service Registration Administration Service Rate-based IPS RC4 (Ron s Code 4) ist eine Verschlüsselungsmethode, die 1987 von Ronald L. Rivest entwickelt wurde Requests for Comments The Relative ID of a Windows Domain Group is the last numeric part of the Domain Group SID (Secure ID). Receive Loudness Rating Restriction of Hazardous Substances Remote PCAP Asymmetric procedure or algorithm for encryption of discrete data, named after its inventors Ronald L. Rivest, Adi Shamir and Leonard Adleman. Realm Specific IP Receive Signal Strength Indicator Rapid Spanning Tree Protocol Resource Reservation Setup Protocol Real-Time Control Protocol Real-Time Transport Protocol RTP Audio Video Profile RTP Early Feedback Profile RTP for Secure Real-Time Communications RTP Based Feedback Frequently asked Questions / Trouble Shooting Page 55

56 RTTTL Ringing Tones Text Transfer Language S S/FTP Screened / Foiled Twisted Pair Cable SAP Service Access Point SAX Simple API for XML SCCP Skinny Call Control Protocol SCP Secure Copy SDP Session Description Protocol SELV Safety Extra Low Voltage SF/FTP Screened Foiled / Foiled Twisted Pair Cable SFTP Secure (SSH) File Transfer Protocol SHA Secure Hash Algorithm SID Windows Domain Group Secure ID SIEM Security Information and Event Management SIF Safety Instrumented Function SIL Safety Integrity Level SIP Session Initiation Protocol SIPS Session Initiation Protocol Secure SLA Service Level Agreement SLR Send Loudness Rating SMI-S Storage Management Initiative Specification SMTP Simple Mail Transfer Protocol SNIA Storage Networking Industry Association SNMP Simple Network Management Protocol SNTP Simple Network Time Protocol SOA Start of Authority Record SOAP SOAP (originally defined as Simple Object Access Protocol) is a protocol specification for exchanging structured information in the implementation of web services in computer networks. SQL Structured Query Language SRTCP Secure Real-Time Control Protocol SRTP Secure Real-Time Transport Protocol SRV DNS-RR Service Locator SS7 Signalling System #7 SSH Secure Shell SSL Secure Sockets Layer STI Speech Transmission Index STMR Side Tone Masking Rating STP Shielded Twisted Pair Cable STUN Session Transversal Utilities for NAT SYSLOG SYSLOG is a standard for forwarding log messages in an IP network. Page 56 Frequently asked Questions / Trouble Shooting

57 T T.30 FAX Protocol G3 Standard T.38 FAX Protocol G3 Standard over IP (FoIP) TACACS Terminal Access Controller Control System TACACS+ Terminal Access Controller Control System Plus TAPI Telephony Application Programming Interface TCF Training Check Function (FAX) TCP Transmission Control Protocol TDM Time Division Multiplex Telnet Teletype Network Protocol TFTP Trivial File Transfer Protocol TIA Telecommunication Industry Association TLS Transport Layer Security TLV Type Length Value TNV Telecommunications Network Voltage ToS Type of Service TOSQA Telecommunication Objective Speech Quality Assessment TRCU Technical Regulation of Customs Union TSIP TCP Session Initiation Protocol TURN Transversal Using Relay NAT U U/FTP U/UTP UA UAC UAS UDP UL UMS UMTS URI URL URN UT UTC UTP Unscreened / Foiled Twisted Pair Cable Unscreened / Unshielded Twisted Pair Cable User Agent User Agent Client User Agent Server User Datagram Protocol Underwriters Laboratories Inc. Unified Messaging Universal Mobile Telecommunications Systems Uniform Resource Identifier Uniform Resource Locator Uniform Resource Name Universal Time Universal Time Coordinated Unshielded Twisted Pair Cable Frequently asked Questions / Trouble Shooting Page 57

58 V VAD VB VdS VIP VLAN VoIP VPN W WebDAV WEEE WINS WLAN WSDL Voice Activity Detection Visual Basic Vertrauen durch Sicherheit Very Important Person Virtual Local Area Network Voice over IP Virtual Private Network Web-based Distributed Authoring and Versioning Waste Electrical and Electronic Equipment Windows Internet Name Service Wireless LAN Web Service Description Language X X.509 ITU-T standard for a public-key-infrastructure X.680ff ITU-T notation for ASN.1 X.690ff ITU-T standards for ASN.1 XER XML Encoding Rules (ASN.1) XML Extensible Markup Language XSL Extensible Stylesheet Language XSLT XSL Transformation, short XSLT, is a programming language for the Transformation of XML-Documents. Page 58 Frequently asked Questions / Trouble Shooting

59 12 Notes Frequently asked Questions / Trouble Shooting Page 59

60 Subject to alterations or errors FHF Funke + Huster Fernsig GmbH Gewerbeallee D Mülheim an der Ruhr Web: Phone: +49/208/ Fax: +49/208/ info@fhf.de Orders: fhf-orders@eaton.com Requests: fhf-sales@eaton.com Support: fhf-support@eaton.com