DMR Conventional Radio. SIP Phone Application Notes

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1 DMR Conventional Radio SIP Phone Application Notes

2 Copyright Information Hytera is the trademark or registered trademark of Hytera Communications Corporation Limited (the Company) in PRC and/or other countries or areas. The Company retains the ownership of its trademarks and product names. All other trademarks and/or product names that may be used in this manual are properties of their respective owners. The product described in this manual may include the Company s computer programs stored in memory or other media. Laws in PRC and/or other countries or areas protect the exclusive rights of the Company with respect to its computer programs. The purchase of this product shall not be deemed to grant, either directly or by implication, any rights to the purchaser regarding the Company s computer programs. Any of the Company s computer programs may not be copied, modified, distributed, decompiled, or reverse-engineered in any manner without the prior written consent of the Company. Disclaimer The Company endeavors to achieve the accuracy and completeness of this manual, but no warranty of accuracy or reliability is given. All the specifications and designs are subject to change without notice due to continuous technology development. No part of this manual may be copied, modified, translated, or distributed in any manner without the express written permission of us. We do not guarantee, for any particular purpose, the accuracy, validity, timeliness, legitimacy or completeness of the Third Party products and contents involved in this document. If you have any suggestions or would like to learn more details, please visit our website at:

3 SIP Phone Application Notes Contents Contents Documentation Information Overview Definition Principle Registration Radios Calling Telephone Devices Telephone Calling Radio Typical Network Topology Restriction Version Application Requirements Device Requirements Terminals Network Device Network Requirements Reference Connection and Configuration Connection Connection Diagram Connection Instructions Configuration Configuration Tools Configuring IPPBX Device Configuring the Repeater Radio Setting Communication Procedure Dialing Example Radio Calling Telephone Telephone Calling Radio Dial Scheme Called Radio Number Called Phone Number FAQ i

4 SIP Phone Application Notes Documentation Information Documentation Information Conventions Icon Conventions Icon Tip Note Caution Warning Danger Description Indicates information that can help you make better use of your product. Indicates references that can further describe the related topics. Indicates situations that could cause data loss or equipment damage. Indicates situations that could cause minor personal injury. Indicates situations that could cause major personal injury or even death. Notation Conventions Notation Bold -> Description The quotation marks enclose the name of a software interface element. For example, click OK. The text in boldface denotes the name of a hardware button. For example, press the PTT key. The symbol directs you to access a multi-level menu. For example, to select New from the File menu, we will describe it as follows: File -> New. Revision History Version Release Date Description Added descriptions on the following features: One-key Connect and One-key Disconnect R All-call SIP remote port Applicable to R8.0 1

5 Documentation Information SIP Phone Application Notes Version Release Date Description R Modified the description of SIP extension and SIP extension password. R Added instructions on Dial-up Mapping. Applicable to R7.0 Added IPPBX configuration. Added registration procedure in Principle. R Added radio private contact in repeater CPS programming. Deleted restriction on operation slot of the radio. Added Priority Setting. R Added restriction on operation slot of the radio. R Updated based on R1.0. R Initial release. 2

6 SIP Phone Application Notes Overview 1. Overview 1.1 Definition SIP (Session Initiation Protocol) Phone is a feature complied with SIP protocol standard. This feature takes the repeater as a carrier to realize the real-time communications between the radio and telephones, such as PSTN phones, VoIP phones and mobile phones. Radio referred here covers portable radio and mobile radio. Here the portable radio is taken as the example. SIP is a standard protocol defined by IETF (Internet Engineering Task Force) to set up, terminate and modify interactive user sessions which include multimedia elements such as video, voice, instant communication, online game and virtual reality. Being open and smart, SIP protocol is now widely applied in VoIP as the development of the Internet. SIP Phone can realize the real-time communications between radios and VoIP phones, PSTN phones and mobile phones. SIP Phone feature has the following highlights: Reducing the cost of building and maintaining communication networks. Allowing users to choose appropriate contacts conveniently. 1.2 Principle The repeaters register all radio contacts to the IPPBX device, and then communicate with other telephones via the IPPBX device to realize the real-time communications between radios and telephones Registration After connected with the SIP phone network, the repeater registers the phone call contacts with the IPPBX device upon power-on. Afterwards, the radios can communicate with telephones. Phone call contact covers private contact, group and all call contact which can communicate with the telephone. 3

7 Overview SIP Phone Application Notes Figure 1-1 Repeater Registering Phone Call Contacts Radios Calling Telephone Devices When making a call to the telephone terminal, the phone number input through the DTMF (Dual Tone Multi-Frequency) keypad is generated into a DTMF signaling which will be sent to the repeater. The repeater decodes the received signaling to the number and converts it to an SIP call request to the IPPBX device. The IPPBX device will search for the address and location of the called number, and then access the telephone network via the corresponding interface accordingly to call the number. See below: Telephone Calling Radio Figure 1-2 Diagram of Radio Calling Telephone When making a call to the radio, the telephone network sends the request to the IPPBX device, the IPPBX device forwards the request to the repeater, and then the repeater calls the radio. See below: 4

8 SIP Phone Application Notes Overview Figure 1-3 Diagram of Telephone Calling Radio 1.3 Typical Network Topology The Typical Network topology consists of radios, repeaters, Ethernet switches, IPPBX and telephones. The radios access the Ethernet via the repeaters, then the repeaters connect to IPPBX via the Ethernet, and finally IPPBX connects to the telephones via the telephone network. See below. Figure 1-4 Typical Network Topology The communication capacity depends on the IPPBX and network deployment. In an SIP phone network, multiple repeaters can be connected, PSTN phones and mobile phones can be connected through the RJ11 interface, VoIP phones the RJ45 interface, and moreover, cascade of multiple IPPBX can form an even bigger network to realize the communication between radios, PSTN phones, VoIP phones and mobile phones. 5

9 Overview SIP Phone Application Notes 1.4 Restriction The repeater must comply with the protocol SIP/2.0(RFC 3261). In the IP Multi-site Connect system, to avoid a heavy load of the repeater, it is not advised to enable SIP feature on the dispatcher or repeater with the third-party service enabled. This feature is available to conventional radios and repeaters on digital channel only. The conventional radio using CPS-ZJ as the dialing rule doesn t support SIP. This feature is available to portable radios and mobile radios with display only. SIP phone is incompatible with AIS on the same repeater or IP Multi-site Connect system. The parameter Telephone Interconnection Enable must be checked via CPS for the repeater. Please refer to Configuring the Repeater for detailed configuration. Only when Phone System is enabled on the radio channel, exactly, the parameter Phone System is not set to None, the radio can initiate phone calls. If the firmware version of the radio is lower than R6.5, the channel parameter TX Admit must be set to Always Allow. Please refer to Radio Setting for detailed configuration. If the parameter Third Party Connect Mode is set to Normal (CPS configuration path: Conventional -> General Setting -> Network -> Application Programming Interface), then only one repeater in the system can enable Telephone Interconnection, otherwise the communications may be terminated abnormally. If Third Party Connect Mode is set to Selective, all repeaters in the system must enable this feature. If the firmware version of the repeater is R6.0 or R6.5, it is recommended to set Third Party Connect Mode to Normal. From firmware version R7.0 and above, the Phone feature is applied to the whole network by default, exactly, only one repeater is enabled with Phone feature, all radios in the system will communicate with telephone system via this repeater. The phone gate ID, radio ID and repeater ID must be unique from each other. 1.5 Version Repeaters and radios with firmware version R8.0 and above support One-Key Connect and One-Key Disconnect. Repeaters and radios with firmware version R7.0 and above support Dial-up Mapping feature. From R6.5 and above, the registration method of SIP Phone extension number is changed from via RRS service of R6.0 to the repeater. Repeaters and radios with firmware version R6.0 and above support SIP Phone feature. For more upgrade instructions on SIP Phone feature, please refer to the corresponding Release Notes. 6

10 SIP Phone Application Notes Application Requirements 2. Application Requirements 2.1 Device Requirements Terminals Radio DMR conventional radios with display, such as PD780, MD780 and X1p, are included. PD780 is taken to introduce the configuration hereinafter. Repeater Generally, DMR repeaters, e.g., RD980S, are employed. RD980S is taken to introduce the configuration hereinafter. Telephone Please select accordingly Network Device Switch Include Ethernet switch, optical fiber switch, and phone call switch. Please consult the supplier for detailed information. Router Device Include firewall, NAT and router, e.g. CISCO Please consult the supplier for detailed information. IPPBX Device IPPBX Device is a dedicated IP switch which provides data switch services for different physical interfaces. It is recommended to use Grandstream ( TEL: , Support TEL: ) UCM Network Requirements A telephone network provided by Telecommunication Operator or a dedicated telephone network is needed. IP network. 7

11 Reference SIP Phone Application Notes 3. Reference N/A 8

12 SIP Phone Application Notes Connection and Configuration 4. Connection and Configuration 4.1 Connection Connection Diagram The repeaters are connected to the IPPBX device via one or more switches first, and then they are connected to the telephone devices via the IPPBX device. In this way, multiple communication devices on different locations are connected together to build an SIP phone system. In this system, radios, PSTN phones and VoIP phones can communicate with each other. Figure 4-1 Device Connection 9

13 Connection and Configuration SIP Phone Application Notes Connection Instructions The radio and repeater transfer data to each other through air interface protocol. IPPBX and PSTN phone are connected through the RJ11 interfaces (FXO interface for external public telephone; FXS interface for internal telephone) and transfer voice data through the RJ11 cable. IPPBX and VoIP phone are connected through the RJ45 interface. Other devices are connected through RJ45 interface and transfer voice data through network cable. Please refer to corresponding references or consult the device operators for detailed information of different devices. 4.2 Configuration Configuration Tools Please set SIP Phone parameters according to network topology and actual requirements. Tools for configuration are listed below: CPS The radio and repeater are configured via Customer Programming Software (CPS). The CPS version must be R7.0 or above. You can consult the manufacturer for more information on CPS. For better configuration, please refer to the help file of CPS for details Configuring IPPBX Device To make sure the radio and telephone device can access the telephone network properly, please configure the following parameters according to the reference provided by the device supplier. SIP extension: Set the SIP phone extension number of private call, group call and all call contacts. The number of SIP extension must be the same as the Phone ID in the Phone Call List of the repeater. Each radio is considered to be an extension and registers with the IPPBX through the repeater. Caution For forming rules of radio private call, group call and all call contacts configured in SIP extension, please see Phone Call Configuration. The private call, group call and all call contacts in the Phone Call List (See Phone Call Configuration) of the repeater must be added to the SIP extension; otherwise, the phone call will failed. 10

14 SIP Phone Application Notes Connection and Configuration SIP extension password: Set the code for SIP extension to access IPPBX. It corresponds to the PBX Access Code of the phone configuration interface of the repeater (CPS Configuration Path: Conventional -> Phone -> Phone System -> Phone System N -> PBX Access Code). Caution In one telephone system, the SIP passwords of all SIP extensions registered with IPPBX via the repeater must be the same, otherwise the contacts cannot be registered properly. IP address of WAN/LAN: The repeater connects to the IPPBX device through this IP address. This parameter is corresponding to the Telephone Gateway IP parameter of the repeater (CPS Configuration Path: Conventional -> Phone -> Phone System -> Phone System N -> Telephone Gateway IP). When connecting to the IPPBX through WAN port, you can acquire the IP address of the IPPBX from the LCD of the IPPBX; When connecting to the IPPBX through LAN port, the IP address of the IPPBX must be set properly (UCM Configuration Path: Settings -> Network Settings -> Basic Settings -> LAN), and you can also acquire the IP address of the IPPBX from the LCD of the IPPBX. Keep-alive: It is recommended that the Keep-alive of IPPBX devices should be disabled to improve the stability of the link. Here Grandstream UCM 6102 is taken as the example. In the following example, only the parameters which must be configured will be described. For more parameter details, please refer to the corresponding configuration guide of Grandstream. Step 1 Access web configuration screen of UCM via browser. The web address of UCM is IP is the IP address of WAN/LAN mentioned above as well as UCM; 8089 is the default port of UCM. The default administrator account and password are both admin. Step 2 In the UCM interface, go to PBX->Basic/Call Routes->Extensions->Create New User to add a new extension. Set Extension and SIP/IAX Password properly in Create New User page, check SIP and clear Enable Keep-alive, and then click Save. 11

15 Connection and Configuration SIP Phone Application Notes Figure 4-2 Adding a New Extension The number of Extension cannot be out of range, and it must be the same as the Called Number in 6.1Called Radio Number; otherwise, the call will fail. The extension number range can be viewed and modified in General page (path: PBX->Internal Options-> General -> Extension Preference -> User Extension). Step 3 In the UCM interface, go to PBX->Basic/Call Routes ->Analog Trunks->Create New Analog Trunk to create an analog trunk. Caution: If you want to connect PBX with an RJ11 cable, you need to create an analog trunk via this procedure. If not, you can skip this procedure. In the page below, select the Channels as per actual needs and enter the analog trunk name in Trunk Name, and then click Save. 12

16 SIP Phone Application Notes Connection and Configuration Figure 4-3 Creating an Analog Trunk Step 4 In the UCM interface, go to PBX->Basic/Call Routes ->VoIP Trunks->Create SIP/IAX Trunks to create a VoIP trunk. Caution: If you want to connect PBX with a network cable, you need to create a VoIP trunk via this procedure. If not, you can skip this procedure. Enter the IP address of upper level PBX in Host Name and enter a valid extension number and password assigned by this PBX. For example, input 2126, and leave the Password blank, and then click Save. 13

17 Connection and Configuration SIP Phone Application Notes Figure 4-4 Creating a VoIP Trunk Step 5 In the UCM interface, go to PBX->Basic/Call Routes ->Outbound Routes->Create New Outbound Rule to create the outbound rule. Please configure the Pattern according to the digits of the extension number. Generally, the extension number has four digits, thus, the Pattern can be set to _XXXX. If you want to make a call to the mobile phone, you need to supplement the Pattern of mobile phone number, for example, _XXXXXXXXXXX. Select the created analog trunk or VoIP trunk in the Use Trunk. Enter 0 in Strip, which allows you to make a phone call directly, and then click Save". Caution: If PBX is connected with the RJ45 cable as well as the RJ11 cable, you need to create two outbound rules for analog trunk and VoIP trunk respectively. 14

18 SIP Phone Application Notes Connection and Configuration Figure 4-5 Creating an Outbound Rule Step 6 In the UCM interface, go to PBX->Call Features->IVR->Create New IVR to create an IVR, and then click Save. 15

19 Connection and Configuration SIP Phone Application Notes Figure 4-6 Creating an IVR Step 7 In the UCM interface, go to PBX->Basic/Call Routes ->Inbound Routes->Create New Inbound Rule to create the inbound rule. Select the created analog trunk or VoIP trunk in the Trunks and select the created IVR in Default Destination, and then click Save. Caution: If PBX is connected with the RJ45 cable as well as the RJ11 cable, you need to create two outbound rules for analog trunk and VoIP trunk respectively. 16

20 SIP Phone Application Notes Connection and Configuration Figure 4-7 Creating an Inbound Rule Note If the SIP remote port is not the default 5060, you need to go to PBX->SIP Settings->Common on the UCM interface to change the Binded UDP Port to the actual SIP remote port, and then click Save Configuring the Repeater Phone System Parameter Configuration Path: Conventional -> Phone -> Phone System -> Phone System N. See Figure 4-8 Phone System Configuration. Parameters: See the parameters in Figure 4-8 Phone System Configuration. Description: See Table 4-1 Phone Configuration. Caution: The Connect Code and Disconnect Code must be different; otherwise, the repeater cannot access the phone system properly. 17

21 Connection and Configuration SIP Phone Application Notes Figure 4-8 Phone System Configuration Parameter Description Setting Telephone Interconnection Enable Telephone Gateway IP Set whether enable the Telephone Interconnection Enable feature. Set the IP address of the IPPBX device. This address must be with the same as that of the IP address of LAN of the IPPBX device; otherwise the repeater cannot be connected to IPPBX. Method: Check Method: Manual input Range: Radio Service Port Voice Slot1 The port which the repeater uses when transferring telephone voice services in Slot 1 or 2. Make sure the port number is unique and even. Method: Manual input Range:

22 SIP Phone Application Notes Connection and Configuration Parameter Description Setting Radio Service Port Voice Slot2 Set the authentication code which the repeater uses to register the called contact information with the IPPBX device. PBX Code Access This value must be with the same as that of the SIP extension password of the IPPBX device; otherwise the called contact cannot be registered properly. Method: Manual input Range: 32-digits ASCII characters If there is no IPPBX device extension password, leave blank here. Phone Gateway ID Wait PBX ACK Timer Radio De-key Beep Enable Set the ID which the repeater uses to identify the current call as a phone call. Set the time in which the repeater will wait for ACK from the telephone after the radio initiates a SIP phone call via the repeater. If the repeater cannot receive ACK from the telephone before the time expires, the call fails. Please set the time according to actual situation. For example, it takes a comparatively longer time for the telephone to connect with the external phone network via IPPBX; in this case, the time should be prolonged accordingly. When this item is checked, the phone user will hear a beep when the radio user releases the PTT during a call. Method: Manual input Range: Method: Manual input Range: 1-10 seconds Method: Check TOT Time Phone With this feature enabled, the phone user will hear a beep when the radio user releases the PTT during a call. Method: Manual input Range: Infinite, minutes 19

23 Connection and Configuration SIP Phone Application Notes Parameter Description Setting Set the number which the repeater uses to identify phone call answering status of the radio. Number (Connect Code) Number (Disconnect Code) You can send the Connect Code by holding down the PTT key. The repeater identifies phone call answering status of the radio according to the Connect Code. Set the number which the repeater uses to identify phone call rejection status of the radio. You can send the Disconnect Code by holding down the PTT key. The repeater identifies phone call rejection status of the radio according to the Disconnect Code. Table 4-1 Phone Configuration Method: Manual input Range: 0-9 (whole number), A, B, C, D, *, # Phone Call Configuration Path: Conventional -> Phone -> Phone Call -> Phone Call List. See Figure 4-9. Parameters: See the parameters in the orange circles in Figure 4-9. Description: Set the Phone Call ID and the Slot ID which the repeater uses to forward the Phone Call. Phone call contacts indicate the private, group and all call contacts which can communicate with telephones. Please refer to the CPS help file for parameter description. When configuring the phone call contacts, the phone ID must be unique; otherwise the phone ID cannot be registered properly. Note When Keep-alive option is enabled for IPPBX, it is recommended to keep the number of phone call contacts capped at 32; when this option is disabled, the maximum number of phone call contacts is recommended to be capped at 64. After the SIP phone network is connected, the repeater will register the set Phone Call information with the IPPBX device upon power-on. When the phone makes a call to the Phone Call ID, please refer to 6.1 Called Radio Number for proper dial scheme; otherwise, the call failed. 20

24 SIP Phone Application Notes Connection and Configuration The slot set here must be with the same as the current one used by the radio. For example, if the slot here is Slot 1 while the radio used Slot 2, the radio cannot receive the call. If the slot of the radio is Pseudo Trunk, the slot here can be Slot 1 or Slot 2 and the radio still can receive calls. Figure 4-9 Phone Call Configuration Priority Setting Path: Conventional -> General Setting -> Accessories -> Priority Control. See Figure 4-10 and Figure Parameters: See the parameters in the orange circles in Figure 4-10 and Figure Description: To ensure normal phone call service, it is recommended to enable Phone Priority option. Figure 4-10 Priority Control for Repeater enabling Phone Call feature If a repeater not enabling the Phone Call feature needs phone call service, you need to set Path Priority to Repeat Request and Repeat Request Priority Local Repeating. 21

25 Connection and Configuration SIP Phone Application Notes Figure 4-11 Priority Control for Repeater not enabling Phone Call feature Radio Setting Phone System Parameter Configuration Path: Conventional -> Phone -> Phone System -> Phone System N. See Figure Parameters: See the parameters in Figure Description: Please refer to Table 4-2 for key parameters of Phone System and refer to the CPS help file for other parameter descriptions. Caution: The Connect Code and Disconnect Code must be different; otherwise, the radio cannot access the phone system properly. 22

26 SIP Phone Application Notes Connection and Configuration Figure 4-12 Phone System Configuration Interface for Radio Parameter Description Setting 23

27 Connection and Configuration SIP Phone Application Notes Parameter Description Setting Set the DTMF Tx Gain in digital channel. The higher the gain value is, the stronger the transmitted DTMF Digital Tx Gain DTMF signal is. When the radio employs AMBE+2 TM Audio Codec Technology, the value of this parameter must be higher than 4 for the radio to support the extension dialing or making phone call with password properly. Method: Manual input Range: -8-8 db Phone Gateway ID Buffer Dial Contact Name Set the ID which the radio uses to identify the current call as a phone call. This parameter must be the same as that of the repeater (See Configuring the Repeater) for the repeater to identify phone calls properly. Set the type of Buffer Dial Contact Name. Follow Tx Contact Name: Save the Tx Contact Name preset for the channel. Gateway ID: Save the Phone Gateway ID. Method: Manual input Range: Method: Select from the dropdown list. Note: To make phone calls, this parameter must be set to Gateway ID Button (Connect Code) Set the button to quickly view the Connect Code of the radio. When the radio receives a phone call, you can press this button to access the DTMF dial box. After inputting the preset Connect Code in the box, hold down the PTT key to answer the call. None (Radio user must input the Connect Code via the keypad manually) P1 (for portable radio with display only) P5 (for mobile radio only) Method: Select from the drop-down list. Note: The button is valid only in DTMF keypad mode. One-key Connect Set whether enable One-key Connect. When this feature is enabled you can press the configured button to send the connect code to enter the system. Method: Check 24

28 SIP Phone Application Notes Connection and Configuration Parameter Description Setting Set the code for the radio to access the phone Number (Connect Code) system. When you send the Connect Code by holding down the PTT key, the repeater identifies phone call answering status of the radio according to the Connect Code. This number must be with the same as that of the repeater (See Configuring the Repeater). Method: Manual input Range: 0-9 (whole number), A, B, C, D, *, # Note: A, B, C, D are not available for radio keypad inputting now. Button (Disconnect Code) Set the button to quickly view the Disconnect Code of the radio. In case of an incoming call or during an ongoing call, you can press this button to access the DTMF dial box. After inputting the preset Disconnect Code in the box, hold down the PTT key to reject the incoming call or hang up the ongoing call. None (Radio user must input the Disconnect Code via the keypad manually) P2 (for portable radio with display only) P6 (for mobile radio only) Method: Select from the dropdown list. Note: The button is valid only in DTMF keypad mode. One-key Disconnect Set whether enable One-key Disconnect. When this feature is enabled you can press the configured button to send the disconnect code to exit the system. Method: Check 25

29 Connection and Configuration SIP Phone Application Notes Parameter Description Setting Set the code for the radio to reject to access or exit Number (Disconnect Code) the phone system. When you send the Disconnect Code by holding down the PTT key, the repeater identifies phone call rejection or termination status of the radio according to the Disconnect Code. This number must be with the same as that of the repeater (See Configuring the Repeater) for the repeater to identify phone call status properly. Method: Manual input Range: 0-9 (whole number), A, B, C, D, *, # Note: A, B, C, D are not available for radio keypad inputting now. Table 4-2 Phone System Key Description for Radio Configuration of Phone List Path: Conventional -> Phone -> Phone List. See Figure Parameters: See the parameters in the orange circles in Figure Description: Set the alias and number of the phone call contact. You can view and make phone calls to the preset Phone Call Alias and corresponding numbers via the radio menu. Please refer to the CPS help file for parameter description. Figure 4-13 Phone List Configuration 26

30 SIP Phone Application Notes Connection and Configuration Channel Configuration Path: Conventional -> Channel -> Digital Channel -> CH Dn. See Figure Parameters: See the parameters in the orange circles in Figure Description: See Table 4-3. Figure 4-14 Channel Configuration Parameter Description Setting Method: Select Slot 1 or Slot 2 in the drop-down list. Slot Operation Set the slot for communication or data transferring. Note: Pseudo Trunk is not supported now. If it is selected, the radio may fail to call the telephone. RRS Channel Revert Set the channel for Radio Register Service (RRS). When the radio registers with the RRS server, it reverts to this channel for registration. After the registration, the radio goes back to work on the previous channel. Method: Select Selected from the drop-down list. 27

31 Connection and Configuration SIP Phone Application Notes Parameter Description Setting Method: Select from the drop-down list. Phone System Set a preset Phone System for the digital channel. When the radio operates in that channel, you can use the Phone System set for the channel. Note: You can configure the Phone System in the drop-down list via Phone -> Phone System. Please refer to Phone System Parameter Configuration. Table 4-3 Phone System on Channel Description Phone Menu Configuration Path: Conventional -> General Setting -> Menu -> Common Menu -> Phone. See Figure Parameter: Phone Description: Set whether include Phone on the radio menu. You can access the Phone menu via the menu. Figure 4-15 Phone Menu Configuration Accessing the Phone Menu: After configuration, you can access the Phone menu using the following methods: (the digital channels of PD78X is taken as the example) Caution: You can access the Phone menu only when it is checked and Channel Phone System is configured. 28

32 SIP Phone Application Notes Connection and Configuration 1. In the home screen, go to Menu -> Phone. Figure 4-16 Main Menu 2. In the Phone menu, you can view the Phone List or input a phone number via Manual Dial. Note Figure 4-17 Phone Menu Phone List saves private contacts preset via CPS. Please refer to Phone Call Configuration. DTMF Keypad Programmable Keys (Optional) Path: Conventional -> General Setting -> Buttons. See Figure Parameter: See the parameters in the orange circles in Figure Description: After any of the following keys is programmed with DTMF Keypad feature, you can press this key to enable or disable the function. When it is enabled, you can input the phone number via the keypad in the home screen, and then the Connect Code button and Disconnect Code button are valid. Note You can also enable the DTMF Keypad feature by going to Menu -> Phone -> DTMF Keypad. 29

33 Connection and Configuration SIP Phone Application Notes Figure 4-18 DTMF Keypad Programmable Keys Configuration 30

34 SIP Phone Application Notes Communication Procedure 5. Communication Procedure 5.1 Dialing Example The radio calls to the telephone by accessing the telephone network and uses the dialing rules specified by the telephone network. Thus, the call procedure for the radio and telephone to call each other is the same. In this case, the radio can be considered as a telephone. When the telephone and the called radio are within a LAN, the calls between them are internal calls. Otherwise, the calls are trans-regional calls. For example, the dialing rule of Company A is as follows: Dial the internal number directly to make an internal call. Input 0 before the external number to make a trans-regional call. Now, For the radio to call the internal telephone, input the extension number of the telephone and hold down the PTT key to initiate a call. For the radio to call the external telephone (including the mobile phone), input 0 before the number and hold down the PTT key to initiate a call. For the internal telephone to call the internal radio, input the radio number to make a call directly (for Dial Scheme, please refer to 6.1 Called Radio Number). For the external telephone to call the internal radio, dial the phone number of Company A first, and then input the called number following the instructions to make a call. 5.2 Radio Calling Telephone The procedure of a radio calling a telephone is as follows: Caution: Radios can only make private call rather than group call to telephone. But the telephone can make private, group or all call to the radios. Do not exit the DTMF Keypad mode during the calling process. Precondition: Press the programmed DTMF Keypad key or go to Menu -> Phone -> DTMF Keypad to enable the DTMF Keypad mode. The radio will display the DTMF Key icon in the home screen. Step 1 Input the phone number (for Dial Scheme, please refer to 6.2 Called Phone Number) or select a preset contact. Do as follows: 31

35 Communication Procedure SIP Phone Application Notes 1. Input the phone number using the numeric keypad directly in the home screen 2. Go to Menu -> Phone -> Manual Dial and input the phone number 3. Go to Menu -> Phone -> Phone List and select a preset contact Note The radio supports two dialing methods: Buffer Dial and Live Dial. Buffer Dial is taken as the example. Buffer Dial: Similar to the dialing on the mobile phone. Input the complete phone number through the keypad, and then hold down the PTT key to call. Live Dial: Input the complete phone number through the keypad when holding down the PTT key. The radio will make a call to the input number in several seconds (this time is configured via CPS). Step 2 Hold down the PTT key to initiate a phone call to the input number or preset contact. Note Here the Buffer Dial is taken as the example. You can also employ the Live Dial. Step 3 The radio waits the called phone to answer. The called phone will give incoming call alerts. You may be required to input the extension number or account password during call establishment. Please input the numbers via the DTMF keypad following the instructions. Step 4 When the called party answers the call, the call is established successfully. Step 5 Hold down the PTT key to talk. During communication, the radio is operating in simplex mode. It cannot receive the voice from the repeater if you hold down the PTT key to transmit impolitely when the radio is under receiving. Step 6 When the communication is done, you can press the Disconnect Code button and hold down the PTT key to send the preset Disconnect Code to terminate the call. Note If the Disconnect Code button is not programmed, you have to enter the Disconnect Code via the numeric keypad manually and then hold down the PTT key to send it. The radio cannot terminate the group or all call initiated by the phone, but can stop receiving by switching to another channel. 32

36 SIP Phone Application Notes Communication Procedure 5.3 Telephone Calling Radio The procedure of a telephone calling a radio is as follows: Step 1 Pick up the phone. Step 2 Input the number (for Dial Scheme, please refer to 6.1 Called Radio Number). Caution: The radio cannot receive phone calls when it is on an analog channel. Step 3 The telephone waits the radio to answer. Step 4 The radio gives alerts. The radio answers the call from the phone automatically. When your radio receives a private call, you can perform alternatively as follow: Press the Connect Code button and hold down the PTT key to send the preset Connect Code to answer the call. Press the Connect Code button to answer the call if One-key Connect is enabled. Press the Disconnect Code button and hold down the PTT key to send the preset Disconnect Code to reject the call. Press the Disconnect Code button to reject the call if One-key Disconnect is enabled. Step 5 When the radio answers the call, the call is established successfully. The call is terminated if the radio rejects to answer. During the call, the telephone operates in duplex mode. Step 6 When the talk is over, the call initiator can end the call by hanging up. 33

37 Dial Scheme SIP Phone Application Notes 6. Dial Scheme 6.1 Called Radio Number When the telephone calls the radio, phone user inputs the Called Number using the numeric keypad directly. If Dial-up Mapping feature is not enabled, the dial scheme of the Called Number is as follows: Called Number = Call Type + Slot# + Target ID Parameter Description Value Range Set the type of calls, namely the Call Type under Phone Call Type Slot# Target ID Call Configuration. 1 indicates private call, 2 group call and 3 all call. Set the slot in which the repeater transmits and receives voice signals, namely the Slot ID under Phone Call Configuration. 1 indicates Slot1 and 2 Slot2. The private or group call ID of the radio, namely the Target ID under Phone Call Configuration. This ID is related to the call type. Table 6-1 1, 2 or 3 1 or Caution: The Called Radio Number must be consistent with the SIP extension number configured on the IPPBX device. If Dial-up Mapping feature is enabled, the dial scheme of the Called Number is as follows: Called Number = Phone ID Parameter Description Value Range Set the number of the radio for the phone call, exactly, the Phone ID Phone ID in phone contact list. The Phone ID must be unique and one of the

38 SIP Phone Application Notes Dial Scheme Parameter Description Value Range corresponding Radio ID, Slot# and Call Type of each Phone ID must be unique. That is to say, at least one of the Radio ID, Slot# and Call Type between two Phone IDs must be unique. With Dial-up Mapping feature enabled, if you want to keep the original dial scheme and extension number of IPPBX, you just need to set the Phone ID on the repeater according to the original dial scheme, that is to set the Phone ID to Call Type + Slot# + Radio ID. Table 6-2 For example, the Radio ID, Slot# and Call Type of Radio A is 3001, Slot1 and Private Call respectively. If Dial-up Mapping is not enabled, you can only make a call to Radio A by dialing If Dial-up Mapping is enabled, you can make a call to Radio A by dialing the Phone ID. For example, if extension number of the Phone ID of Radio Ais 23, you can dial 23 to call Radio A. If you want to keep the previous extension number , you can set the Phone ID to , then you need to dial to call Radio A. 6.2 Called Phone Number When calling a telephone with a radio, you can input the number using the numeric keypad directly. Please refer to 5.1 Dialing Example for detailed dial schemes. 35

39 FAQ SIP Phone Application Notes 7. FAQ Q: Why does the radio give no response to the call from a telephone, and the telephone give alert that the number you dialed is unreachable, please check the number and dial later? A: The radio registration may be failed. Please check the repeater PBX Connect Code and SIP extension password set on IPPBX, they must be the same. Q: Why does the call end automatically when the radio calls the telephone? A: The radio registration may be failed. See the answer above for solutions. Q: Why does the radio keep ringing after inputting the connect code when receiving a telephone call? A: The Connect Code may be incorrect. Please check whether the connect code of the radio is the same as that of the repeater. Or, the Path Priority is not Repeat Request or Repeat Request Priority is not Local Repeating. Q: Why cannot the radio terminate the phone call? A: The radio cannot terminate a group call. For a private phone call, because the Disconnect Code of the radio is not the same as that of the repeater. Q: Why does the radio fail to dial through when trying again? A: Two potential reasons: 1. Network delay. Please check the network. 2. The value of Digital DTMF Tx Gain is lower than 4. Q: Why does the telephone rings again when it hangs up before the radio answers the call? A: The IPPBX detection may be not accurate. Please hang up again. Q: Why cannot the telephone hear the radio while the radio can hear the telephone on a group call initiated by the telephone? A: The Third Party Connect Mode of the repeater is set to Selective. In this mode, the repeater does not 36

40 SIP Phone Application Notes FAQ forward the voice signal to the telephone. Q: Why cannot the radio terminate the call to the telephone? A: The configuration on the request priority of the home repeater of the radio may be incorrect. If the home repeater does not support the Phone feature, the radio can make a phone call through other available repeaters via the IP multi-site connect. In this case, the Repeat Request Priority of the home repeater must be set to Local Repeating. CPS configuration path: Edit -> Conventional -> General Setting -> Accessories -> Priority Control -> Repeat Request Priority. 37

41 Hytera Communications Corporation Limited Hytera Communications Corporation Limited.

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