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� Frequently asked questions

� Trouble-shooting

VoIP-telephones

FHF FAQ-TS-IP_Phones 03/13 V9

Frequently asked Questions / Trouble Shooting

FAQ-TS-IP_Phones-en 03/13 V9 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.

� Note

Please read the operating manual carefully before installing a device.

Copyright © 2013 FHF Funke + Huster Fernsig GmbH Gewerbeallee 15 – 19 45478 Mülheim an der Ruhr Tel +49 (208) 8268 - 0 Fax +49 (208) 8268 - 377 http://www.fhf.de


Table of Contents

1 Frequently asked Questions........................................ 5

1.1 Administration............................................................. 5

1.2 Structure of the Software Version Information.......... 5

1.2.1 Firmware Format (Hotfix Release) ...................................... 5

1.2.2 Firmware Format (Development Release) ........................... 6

1.2.3 Bootcode Format (Hotfix Release) ...................................... 6

1.2.4 Bootcode Format (Development Release) ........................... 7

1.3 Software Update ......................................................... 7

1.3.1 Software Update under Retention of the Main Version of the Software........................................................................... 7

1.3.2 Software Update from V6 to V7 or higher ........................... 8

1.3.3 Software Update from V7 to a higher version...................... 8

1.3.4 Bootcode and Firmware Update ......................................... 8

1.4 Setup of the Language of the User Interface of the Telephone.................................................................... 8

1.4.1 Setup of the Language via DHCP Server ............................ 9

1.4.2 Setup of the Language at the Telephone ............................ 9

1.4.3 Setup of the Language with the Web Interface ................... 9

2 Troubleshooting ........................................................ 11

2.1 Commissioning.......................................................... 11

2.2 Registration............................................................... 12

2.3 Making Calls .............................................................. 14

3 Specifics of PBX’s ...................................................... 15

3.1 3CX (Software PBX) .................................................. 15

3.2 Cisco .......................................................................... 15

3.2.1 Call Manager Express ...................................................... 15

3.2.1.1 Block Dialing ................................................................... 15

3.2.2 Unified Communications Manager (Call Manager) .............. 15

3.2.2.1 Block Dialing ................................................................... 15

3.2.2.2 Configuring Unified Communications Manager for Non-Cisco SIP Phones ..................................................................... 16

3.2.2.3 How Cisco Unified Communications Manager identifies a Third-Party SIP Phone ..................................................... 17

3.2.2.4 Third-Party SIP Phones and TFTP..................................... 17

3.2.2.5 Enabling Digest Authentication for Third-Party SIP Phones. 18

3.2.2.6 Third-Party SIP Phone Configuration Checklist................... 18

3.2.2.7 DTMF Reception.............................................................. 20

3.2.2.8 Licensing Third-Party SIP Phones ..................................... 20

3.2.2.9 Where to find more Information....................................... 22

4 Device internal Troubleshooting............................... 23

5 Wireshark .................................................................. 24

5.1 General ...................................................................... 24


5.2 RPCAP Server und Microsoft Windows ..................... 24

5.2.1 Configuration of the RPCAP Server ................................... 25

5.2.2 Capturing with the Program Wireshark ............................. 25

5.2.2.1 Older Wireshark Versions................................................. 25

5.2.2.2 Newer Wireshark Versions ............................................... 25

6 Abbreviations ............................................................ 26

7 Notes ......................................................................... 34


1 Frequently asked Questions

1.1 Administration

1. In order to set up the VoIP telephone via the Web interface, you must login with a user name and a password.

• In the initial condition the details for user name and password you can find in the manual of the respective device.

1.2 Structure of the Software Version Information

1.2.1 Firmware Format (Hotfix Release)

Old firmware format (hotfix release): [yy-mmhhh.rr]

yy year information

mm main version of the software

hhh main release

rr release

New firmware format (hotfix release): [mmhhh.rr]


hhh main release

rr 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 [9.061001] there are hotfix releases only.


New firmware format V9 (hotfix release): [m.hhhrrr]

m main version of the software

hhh main release

rrr release

1.2.2 Firmware Format (Development Release)

Old firmware format (development release): [yy-mmsss]

yy year information


sss release

New firmware format (development release): [mmsss]


sss release

1.2.3 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 year information


hhh main release

rr release

New bootcode format software V7 and higher (hotfix release): [mmhhhrr]


hhh main release

rr release


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 [9061001] there are hotfix releases only.

New bootcode format V9 (hotfix release): [mhhhrrr]

m main version of the software

hhh main release

rrr release

1.2.4 Bootcode Format (Development Release)

Old bootcode format software V7 and higher (development release): [yy-mmsss]

yy year information


sss release

Neues bootcode format software V7 and higher (development release): [mmsss]


sss release

1.3 Software Update

1.3.1 Software Update under Retention of the Main Version of the Software

The description is available in the manual.


1.3.2 Software Update from V6 to V7 or higher

1. Update the firmware to the hotfix version 08-60900.95 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 an d on again.

5. Update the firmware to the version V7 or a higher Version.


7. Update the bootcodes to the version V7 or a higher version.


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.

1.3.3 Software Update from V7 to a higher version

The description is available in the manual.

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.


1.4.1 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.

1.4.2 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>->Preferences->Language

With the cursor right and cursor left keys a language can se 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.

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.

�

Note

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.


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.


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.

• 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 within 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 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.


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.

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.


3 Specifics of PBX’s

3.1 3CX (Software PBX)

For a correct inquiry call the protocol option SIP-hold (see manual) has to be set to the value 1 (sendonly).

3.2 Cisco

3.2.1 Call Manager Express

3.2.1.1 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:1…6

3.2.2 Unified Communications Manager (Call Manager)

3.2.2.1 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:1…6


3.2.2.2 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.

Feature Setting

Integrated with Centralized TFTP No

Sends MAC Address No

Downloads Softkey File No

Downloads Dial Plan File No

Supports Cisco Unified Communications Manager Failover and Fallback

No

Supports Reset and Restart No

Table 1: Non-Cisco SIP Phone Configuration

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.


3.2.2.3 How Cisco Unified Communications Manager identifies a Third-Party SIP Phone

Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.

The REGISTER message includes the following header:

Authorization: Digest username="swhite",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"

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).

�

Note

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.

3.2.2.4 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.


3.2.2.5 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.

� Note

Cisco Unified Communications Manager does not support Transport Layer Security (TLS) from third-party SIP phones.

3.2.2.6 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 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


Procedure

Step 2 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.

Step 3 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).

Step 4 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.

Step 5 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.

Step 6 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.

Step 7 Add and configure lines (DNs) on the phone.

Step 8 In the End User Configuration window, associate the third-party SIP phone with the user by using Device Association and choosing the SIP phone.

Step 9 In the Digest User field of the Phone Configuration window, choose the end user that you created in Step 3


Procedure

Step 10 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

3.2.2.7 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.

3.2.2.8 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.


�

Note

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.

Field Name Old Value New Value

Maximum Number of Calls Per Device

8 2

Default Maximum Number of Calls per DN

4 2

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

64 16

Default Maximum Number of Calls per DN

4 2

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 DN-related 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.

• 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.

3.2.2.9 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 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.


5 Wireshark

5.1 General

Another possibility to determine protocol problems is the program Wireshark. This open source program is a protocol analyser. It can be downloaded from the internet http://www.wireshark.org 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.

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.

5.2 RPCAP Server und 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 as only available for Windows PCs..

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).

IP telephone

PoE injector

Hub LAN

PC Wireshark

Gatekeeper/ PBX

IP telephone

3

IP telephone

2


5.2.1 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.

5.2.2 Capturing with the Program Wireshark

Start the program Wireshark and open the capture options dialog.

5.2.2.1 Older Wireshark Versions

Enter "rpcap://<IP address of the phone>/trace" into the interface field. Then click the button start to start the capturing.

5.2.2.2 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.


6 Abbreviations

8

802.1X Standard for Authorisation in Computer Networks (IEEE)

802.3af Standard for Power over Ethernet (IEEE)

A

A DNS-RR Address Record

AAAA DNS-RR IPv6 Address Record

AC Alternate Current

ACM Avaya Call Manager

A/D Analogue/Digital

AES Advanced Encryption Standard

ANSI American National Standards Institute

AOR Account of Registration

API Application Program Interface

ARP Address Resolution Protocol

AS-SIP Assured Services-Session Initiation Protocol

ASCII American Standard Code for Information Interchange

ASN.1 Abstract Syntax Notation One

ATEX Atmosphere Explosive

B

BER Basic Encoding Rules (ASN.1)

C

CA Certificate Authority

CCP Compression Control Protocol

CDPN Called Party Number

CDR Call Detail Records

CE Conformite Europeenne

CER Canonical Encoding Rules (ASN.1)

CEST Central European Summer Time

CET Central European Time

CGN Calling Party Number

CGPN Calling Party Number

CLIR Calling Line Identification Restriction

CM Call Manager, see also CUCM

CME Call Manager Express

CNAME DNS-RR Canonical Name Record

CNG Comfort Noise Generation

CPN Calling Party Number


CRC Cyclic Redundancy Check

CRL Certificate Revocation List

CSS Cascading Style Sheets

CSTA Computer Supported Telecommunications Applications

CTI Computer Telephony Integration

CUCM Cisco Unified Communications Manager, see also CM

D

DC Direct Current

DECT Digital Enhanced Cordless Telecommunications

DER File name extension for a base64 coded certificate

DER Distinguished Encoding Rules (ASN.1)

DES Data Encryption Standard

DIN Deutsches Institut für Normung

DHCP Dynamic Host Configuration Protocol

DOM Document Object Model

DN Directory Number

DNS Domain Name Server

DNSBL DNS-based Blackhole List, Block List, or Blacklist

DNS-RR DNS Resource Record

DRAM Dynamic Random Access Memory

DSL Digital Subscriber Line

E

E.164 Standard for the international public telecommunication numbering plan (ITU-T)

EAP Extensible Authentication Protocol

EAR Stiftung Elektro-Altgeräte Register

ECN Encoding Control Notation (ASN.1)

EIA Electronic Industries Alliance

EMC Electromagnetic Compatibility

EMV Elektromagnetische Verträglichkeit

ENUM Telephone Number Mapping

EUI-64 64-bit Extended Unique Identifier


F

F/FTP Foiled / Foiled Twisted Pair Cable

FAT File Allocation Table. A from Microsoft developed file system.

FAT32 A FAT Variant

FCC Federal Communications Commission

FQDN Fully Qualified Domain Name

FQTN Fully Qualified Telephone Number

FTP File Transfer Protocol

FTP Foiled Twisted Pair Cable

G

GMT Greenwich Mean Time

GRE Generic Routing Encapsulation

GSER Generic String Encoding Rules (ASN.1)

H

H.323 Higher recommendation of the ITU for protocols

HTML Hypertext Markup Language

HTTP Hypertext Transfer Protocol

HTTPS Hypertext Transfer Protocol Secure

I

IANA Internet Assigned Numbers Authority

ICANN Internet Corporation for Assigned Names and Numbers

IDN Internationalized Domain Name

IE Internet Explorer

IEC International Electrotechnical Commission

IEEE Institute of Electrical and Electronics Engineers

IEEE 802.3af Standard for PoE

IEEE 802.1X Standard for Authorisation in Computer Networks

IETF Internet Engineering Task Force

IIS Internet Information Server

IP Internet Protocol

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

ISRAM Internal Static Random Access Memory

ITU International Telecommunication Union


J

JAVA Java is a object orientated programming language

JMS Java Message Service

K

Kerberos Kerberos is a computer network authentication protocol

KPML Keypad Markup Language

L

LAN Local Area Network

LCD Liquid Crystal Display

LDAP Lightweight Directory Access Protocol

M

MAC Message Authentication Code

MAC-Address Media Access Control Address

MD5 Message Digest Algorithm 5

MEST Middle European Summer Time

MET Middle European Time

MIB Management Information Base

MoH Music on Hold

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

MTU Maximum Transmission Unit

MX DNS-RR Mail Exchange Record

N

NAPTR DNS-RR Naming Authority Pointer

NAT Network Address Translation

NDP Neighbour Discovery Protocol

NS DNS-RR Name Server Record

NTP Network Time Protocol

O

OBJ Object Identifier

OUI-24 24-bit Organizationally Unique Identifier

OUI-36 36-bit Organizationally Unique Identifier


P

P12 PKCS#12 file (see PFX)

PA Potentialausgleich

PAI P-Asserted Identity

PBX Private Branch Exchange

PC Personal Computer

PCAP PCAP (Packet Capture) free programming interface (API), to capture network traffic.

PDU Protocol Data Unit

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

POSIX Portable Operating System Interface for UniX

PPE Personal Protective Equipment

PPI P-Preferred Identity

PPP Point-to-Point Protocol

PPPoE PPP over Ethernet

PPTP Point-to-Point Tunnelling Protocol

Proxy A hardware server that acts as an intermediary between a work station user and the Internet so that an enterprise can ensure security.

PTB Physikalisch Technische Bundesanstalt

PTR DNS-RR Pointer Record

Q

QoS Quality of Service


R

R&TTE Radio and Telecommunications Terminal Equipment

RADIUS Remote Authentication Dial-In User Service

RAS Registration Administration Service

RC4 RC4 (Ron’s Code 4) ist eine Verschlüsselungsmethode, die 1987 von Ronald L. Rivest entwickelt wurde

RFC Requests for Comments

RID The Relative ID of a Windows Domain Group is the last numeric part of the Domain Group SID (Secure ID).

ROHS Restriction of Hazardous Substances

RPCAP Remote PCAP

RSA Asymmetric procedure or algorithm for encryption of discrete data, named after its inventors Ronald L. Rivest, Adi Shamir and Leonard Adleman.

RSTP Rapid Spanning Tree Protocol

RTCP Real-Time Control Protocol

RTP Real-Time Transport Protocol

RTTTL Ringing Tones Text Transfer Language

S

S/FTP Screened / Foiled Twisted Pair Cable

SF/FTP Screened Foiled / Foiled Twisted Pair Cable

SAX Simple API for XML

SCCP Skinny Call Control Protocol

SCP Secure Copy

SDP Session Description Protocol

SELV Safety Extra Low Voltage

SFTP Secure (SSH) File Transfer Protocol

SHA Secure Hash Algorithm

SID Windows Domain Group Secure ID

SIP Session Initiation Protocol

SIPS Session Initiation Protocol Secure

SNMP Simple Network Management Protocol

SMTP Simple Mail Transfer 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.

SRTCP Secure Real-Time Control Protocol

SRTP Secure Real-Time Transport Protocol

SRV DNS-RR Service Locator

SSH Secure Shell


SSL Secure Sockets Layer

STP Shielded Twisted Pair Cable

STUN Simple Traversal of UDP over NATs

SYSLOG SYSLOG is a standard for forwarding log messages in an IP network.

T

TCP Transmission Control Protocol

Telnet Teletype Network Protocol

TFTP Trivial File Transfer Protocol

TIA Telecommunication Industry Association

TLS Transport Layer Security

TNV Telecommunications Network Voltage

ToS Type of Service

TSIP TCP Session Initiation Protocol

U

U/FTP Unscreened / Foiled Twisted Pair Cable

U/UTP Unscreened / Unshielded Twisted Pair Cable

UDP User Datagram Protocol

UL Underwriters Laboratories Inc.

URI Uniform Resource Identifier

URL Uniform Resource Locator

URN Uniform Resource Name

UTC Universal Time Coordinated

UTP Unshielded Twisted Pair Cable

V

VAD Voice Activity Detection

VB Visual Basic

VIP Very Important Person

VLAN Virtual Local Area Network

VoIP Voice over IP

VPN Virtual Private Network

W

WebDAV Web-based Distributed Authoring and Versioning

WEEE Waste Electrical and Electronic Equipment

WINS Windows Internet Name Service

WLAN Wireless LAN

WSDL 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.


7 Note

Subject to alterations or errors

FHF Funke+Huster Fernsig GmbH

Gewerbeallee 15-19 · D-45478 Mülheim an der RuhrPhone +49 /208 /82 68-0 · Fax +49 /208 /82 68-286

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