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Huawei AR Series: PBX Number Parameters

Huawei AR Series: PBX Number Parameters

When deploying Huawei AR Series Routers to act as an IP PBX, you may need some extra parameters according to your environment. The pbx number-parameter command in Huawei AR routers sets control points for a PBX.

You can apply these parameters in voice view menu, and the number parameter usage format is like below:

pbx number-parameter name value

Here name specifies the name of a control point and the value is an integer that ranges from 0 to 249. Value specifies the value of a control point and is an integer that ranges from 0 to 4294967295.

Since the notation is not easy-readable, you can use the table below for parameters and values:

Number Parameters and Descriptions

Control Point NameDescriptionValue
0Configures whether to support SDP RTCP.Enumerated value. The options are as follows:0: SDP RTCP is not supported.1: SDP RTCP is supported.The default value is 0.
1 to 3Reserved.
4Indicates whether the PBX processes CNG signals.Enumerated value. The options are as follows:0: The PBX does not process CNG signals.1: The PBX processes CNG signals.The default value is 0.
5Sets the delay in processing modem events.The value is an integer that ranges from 0 to 50000, in milliseconds.The default value is 5000. The value 0 indicates that model events are processed immediately.
6Reserved.
7Configures whether to initiate re-negotiation when the H.323 was in early media condition.Enumerated value. The options are as follows:0: Re-negotiation is not initiated.1: Re-negotiation is initiated.The default value is 0.
8Configures whether to use the local or remote priority mode for SDP negotiation.Enumerated value. The options are as follows:0: The local priority mode is used for SDP negotiation.1: The remote priority mode is used for SDP negotiation.The default value is 1.
9Sets the SDP packetization time format.Enumerated value. The options are as follows:0: maxmptime1: ptimeThe default value is 1.
10 to 14Reserved.
15Configures whether to use VBD when G.711 is used.Enumerated value. The options are as follows:0: VBD is not used when the codec is G.711.1: VBD is used when the codec is G.711.The default value is 0.
16Sets the service triggering mode.Enumerated value. The options are as follows:0: Loose coupling mode.1: Semi-tight coupling in inband mode2: Semi-tight coupling in outband mode3: Full coupling modeThe default value is 0.
17Sets the codec mode when the codec priority is 0.Enumerated value. The options are as follows:0: G711U2: G726-324: G7238: G711A9: G72215: G72818: G72920: G723Low21: G726-1622: G726-2423: G726-4096: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 8.
18Sets the packetization time when the codec priority is 0.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 2.
19Sets the codec mode when the codec priority is 1.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 0.
20Sets the packetization time when the codec priority is 1.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 2.
21Sets the codec mode when the codec priority is 2.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 18.
22Sets the packetization time when the codec priority is 2.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 1.
23Sets the codec mode when the codec priority is 3.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 4.
24Sets the packetization time when the codec priority is 3.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 3.
25Sets the codec mode when the codec priority is 4.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 8.
26Sets the packetization time when the codec priority is 4.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 1.
27Sets the codec mode when the codec priority is 5.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 8.
28Sets the packetization time when the codec priority is 5.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 3.
29Sets the codec mode when the codec priority is 6.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 0.
30Sets the packetization time when the codec priority is 6.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 1.
31Sets the codec mode when the codec priority is 7.Enumerated value. The options are as follows:0: G711U2: G7264: G7238: G711A9: G72215: G72818: G72920: G723Low96: R219897: R283398: v15099: clear mode250: T.38251: T.38+2The default value is 250.
32Sets the packetization time when the codec priority is 7.Enumerated value. The options are as follows:0: 5 ms1: 10 ms2: 20 ms3: 30 ms4: 40 ms5: 50 ms6: 60 msThe default value is 2.
33Configures whether to support early media.Enumerated value. The options are as follows:0: Early media is not supported.1: Early media is supported.The default value is 1.
34 to 36Reserved.
37Configures whether to use CLIP polarity reversal.Enumerated value. The options are as follows:0: CLIP polarity reversal is not required.1: CLIP polarity reversal is required.The default value is 0.
38Sets the sequence in which B channels will be selected.Enumerated value. The options are as follows:0: Preferred1: DesignatedThe default value is 0.
39Sets the duration during which the AG0 port detects the calling number.The value is an integer that ranges from 0 to 15, in seconds.The default value is 5.
40Sets the interval between two digits on the AG0 port.The value is an integer that ranges from 0 to 15, in seconds.The default value is 0.
41Sets the period during which a user does not dial any number after picking up the phone.The value is an integer that ranges from 10000 to 30000, in milliseconds.The default value is 10000.
42Sets the interval between two digits for common calls.The value is an integer that ranges from 5000 to 60000, in milliseconds.The default value is 20000.
43Sets Trunk PRI out call Transfer Capability field filled with 3.1 kHz audio or SPEECH.Enumerated value. The options are as follows:0: 3.1 kHz audio1: SPEECHThe default value is 1.
44Sets the duration of waiting for called responseThe value is an integer that ranges from 1500 to 300000, in milliseconds.The default value is 1500.
45Reserved.
46Indicates the waiting time threshold, crossing which an intra-office or local call is regarded as no response.The value is an integer that ranges from 10000 to 300000, in milliseconds.The default value is 60000.
47Indicates the waiting time threshold, crossing which a national toll call is regarded as no response.The value is an integer that ranges from 10000 to 300000, in milliseconds.The default value is 90000.
48Indicates the waiting time threshold, crossing which an international toll call is regarded as no response.The value is an integer that ranges from 10000 to 300000, in milliseconds.The default value is 120000.
49Reserved.
50Configures whether to display the default calling number of the trunk group.Enumerated value. The options are as follows:0: The default calling number is not displayed.1: The default calling number is displayed.The default value is 1.
51 and 52Reserved.
53Sets the CDR format.Enumerated value. The options are as follows:0: CDR1: SOFTX2: MINI3: UCBILLThe default value is 0.
54 to 61Reserved.
62Whether use 91 as clearmode payload.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
63Reserved.
64Specifies whether to check the remote IP address when the H.323 receives an incoming call.Enumerated value. The options are as follows:0: No1: YesThe default value is 1.
65Specifies the duration of the busy tone.The value is an integer that ranges from 1000 to 300000, in milliseconds.The default value is 40000.
66-67Reserved.
68Sets the lifecycle of the CCBS/CCNR service.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 1800.0: The CCBS/CCNR service is always effective.
69Sets the number of CCBS/CCNR service attempts.The value is an integer that ranges from 0 to 20.The default value is 3.0: The CCBS/CCNR service is always effective.
70Sets the interval for CCBS/CCNR service attempts.The value is an integer that ranges from 1 to 1800, in seconds.The default value is 180.
71Whether to delete the pound sign (#) when users call external users.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
72Configures whether the AT0 trunk that does not provide calling number detection releases calls.Enumerated value. The options are as follows:0: The AT0 trunk that does not provide calling number detection does not release calls.1: The AT0 trunk that does not provide calling number detection releases calls.The default value is 1.
73Sets the stream mode for playing the announcement.Enumerated value. The options are as follows:0: Sendonly1: SendRecvThe default value is 0.
74Sets the delay in collecting digits.The value is an integer that ranges from 0 to 1000, in 10 ms.The default value is 100.
75Configures whether to allow user register at local device.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
76Set the dsp ignore event.The value is an integer that ranges from 0 to 62.The default value is 0.
77-83Reserved.
84Sets the FTP server size in the CDR.The value is an integer that ranges from 1 to 30000.The default value is 2400.
85Configures whether to store CDRs locally.Enumerated value. The options are as follows:0: CDRs cannot be stored locally.1: CDRs can be stored locally.The default value is 0.
86Sets the number of stored CDRs.The value is an integer that ranges from 1 to 1024.The default value is 2.
87Sets the interval at which CDRs are saved.The value is an integer that ranges from 1 to 1440, in minutes.The default value is 15.
88Sets the processing policy after the hard disk or memory is full.Enumerated value. The options are as follows:0: Calls are rejected.1: CDRs are not generated.2: Old CDRs are overwritten.The default value is 2.
89 to 93Reserved.
94Sets the MWI mode of the ISDN port.Enumerated value. The options are as follows:0: DEFERRED1: IMMEDIATE2: COMBINEThe default value is 0.
95 to 97Reserved.
98Sets the value of the protection timer.The value is an integer that ranges from 1 to 0xFFFF, in seconds.The default value is 60.
99Reserved.
100Sets the duration for the MWI completes FSK transmission.The value is an integer that ranges from 10 to 0xFFFF, in milliseconds.The default value is 500.
101Sets the MWI format.Enumerated value. The options are as follows:0: The MWI is on or off.1: Defined by the ETSI standards.The default value is 0.
102Configures whether the trunk group sends the number of the minimum length.Enumerated value. The options are as follows:0: The trunk group does not send the number of the minimum length.1: The trunk group sends the number of the minimum length.The default value is 0.
103Configures whether to enable media proxy when the media proxy policy is auto and IP addresses of both communicating parties are the same.Enumerated value. The options are as follows:0: Media proxy is not enabled.1: Media proxy is enabled.The default value is 0.
104Configures whether to enable media proxy when SIP trunks are connected.Enumerated value. The options are as follows:0: Media proxy is enabled when SIP trunks are connected.1: Media proxy is not enabled when SIP trunks are connected.The default value is 1.
105Configures media proxy address selection after IP-side media proxy is enabled.Enumerated value. The options are as follows:0: The configured media IP address is used.1: The IP address between the media IP address and media mapping IP address is used, which is on the same network segment as the remote end.The default value is 1.
106 and 107Reserved.
108Configures whether to use the calling number for the call forwarding service.Enumerated value. The options are as follows:0: The calling number for the call forwarding service is not used.1: The calling number for the call forwarding service is used.The default value is 0.
109 and 110Reserved.
111Configures whether to initiate re-negotiation when the invite200 message sent by the peer end contains multiple codec values.Enumerated value. The options are as follows:0: Re-negotiation is not initiated.1: Re-negotiation is initiated.The default value is 0.
112 and 113Reserved.
114Sets the attributes of the calling or called number over the BRA or PRA trunk.Enumerated value. The options are as follows:0: The call number attribute remains unchanged.1: The international call number attributes are unknown.2: The national call number attributes are unknown.3: The international and national call number attributes are unknown.4: The local call number attributes are unknown.5: The local and international call number attributes are unknown.6: The local and national call number attributes are unknown.7: The local, national, and international call number attributes are unknown.The default value is 0.
115 to 119Reserved.
120Indicates the duration of the T301 timer on the DSS1 side.The value is an integer that ranges from 180 to 65535, in seconds.The default value is 180.
121Indicates the duration of the T302 timer on the DSS1 side.The value is an integer that ranges from 5 to 65535, in seconds.The default value is 10.
122Indicates the duration of the T303 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 4.
123Indicates the duration of the T304 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 20.
124Indicates the duration of the T305 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 30.
125Indicates the duration of the T306 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 30.
126Indicates the duration of the T307 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 180.
127Indicates the duration of the T308 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 4.
128Indicates the duration of the T309 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 90.
129Indicates the duration of the T310 timer on the DSS1 side.The value is an integer that ranges from 0 to 40, in seconds.The default value is 30.
130Indicates the duration of the T312 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 6.
131Indicates the duration of the T314 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 4.
132Indicates the duration of the T316 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 120.
133Indicates the duration of the T317 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 100.
134Indicates the duration of the T320 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 30.
135Indicates the duration of the T321 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 30.
136Indicates the duration of the T322 timer on the DSS1 side.The value is an integer that ranges from 0 to 65535, in seconds.The default value is 4.
137Specifies whether to respond to the session update without reporting the service.Enumerated value. The options are as follows:0: Yes1: NoThe default value is 0.
138Set the IP trunk wait dial number timer length.The value is an integer that ranges from 1 to 15, in seconds.The default value is 5.
139Indicates whether the AR sends subsequent requests to the new address when the peer address or port number is changed in the contact field of the response that the AR receives.Enumerated value. The options are as follows:0: Yes1: NoThe default value is 0.
140Specifies whether to delete the reason header field when calls of FXS or unknown local users are made over the SIP trunk and released.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
141Indicates the timer duration, exceeding which an outgoing local call routed through the SIP trunk is regarded as no response.The value is an integer that ranges from 10000 to 600000, in milliseconds.The default value is 60000.
142Indicates the timer duration, exceeding which an outgoing national toll call routed through the SIP trunk is regarded as no response.The value is an integer that ranges from 10000 to 600000, in milliseconds.The default value is 90000.
143Indicates the timer duration, exceeding which an outgoing international toll call routed through the SIP trunk is regarded as no response.The value is an integer that ranges from 10000 to 600000, in milliseconds.The default value is 120000.
144Set the ISDN trunk wait dial number timer length.The value is an integer that ranges from 1 to 15, in seconds.The default value is 10.
145Reserved.
146Whether to play a congestion tone by DISCONNECT message in the net-to-user direction.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
147Configures whether print the log on the screen.Enumerated value. The options are as follows:0: Yes1: NoThe default value is 0.
148Specifies whether to change the Min-SE value to a valid one if it is 0.Enumerated value. The options are as follows:0: No1: Yes (Change the value to 90)The default value is 0.
149Specifies the aging time of the SIP Flood and DDoS attack prevention whitelist.The value is an integer that ranges from 30 to 1800, in seconds.The default value is 90.
150Specifies whether to send progress to dss1 after alertingEnumerated value. The options are as follows:0: Not send1: SendThe default value is 1.
151Specifies whether to enable limiting after registration fails.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.NOTE:This control point determines whether to start limiting on registration packets only. This control point needs to be used with control point 152 to determine whether to enable limiting on registration packets. The status of this control point does not affect generated entries in the flow limiting table.
152Specifies the failure count before limiting is enabled.The value is an integer that ranges from 1 to 10.The default value is 3.
153Specifies the aging time of entries in the flow limiting table.The value is an integer that ranges from 60 to 1800, in seconds.The default value is 300.NOTE:If the aging time is too short, you can run the debugging voice nm cmd 800 and debugging voice nm cmd 801 p1 0 command on the nm module to perform software commissioning and delete corresponding entries in the flow limiting table.
154Specifies whether to delete the audio line when the media and audio or t38 and audio exist in one media line.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
155Specifies whether the AR plays the ringback tone for the downlink device.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
156Indicates whether a call is released when no sequential ringing group member answers the call.Enumerated value. The options are as follows:0: Calls are not released.1: Calls are released.The default value is 0.
157Indicates whether the SDP-answer that is sent after a media negotiation is performed contains all common codecs shared by the system and SDP-offer.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
158Specifies whether a device immediately sends a Cancel message to release the session with a remote device when it does not receive an Invite response from the remote device after sending an Invite message to the remote external device using a SIP AT0 trunk.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
159Specifies a call waiting service mode.The value is an integer. The options are as follows:0: IVR prompt mode.1: Simplified mode. A user can press the hookflash directly without hearing a prompt tone and switch the call.The default value is 0.
160Specifies whether to convert the Progress messages of a BRA trunk to the Alerting messages.The value is an integer. The options are as follows:0: No1: YesThe default value is 0.
161Specifies whether the IMS core can use UA profiles in SIP to set the service attributes of a PBX user through an SIP AT0 trunk.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
162Indicates whether authentication information is carried in the registration refresh messages sent by AR routers in SIP AT0 and SIP PRA connection scenarios.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
163Indicates whether to forcibly add SDP to 180 Ringing messages that are mapped to the SIP side after an AR receives Alerting messages sent from the PRA side in the scenario where SIP trunk incoming calls are transferred to a PRA trunk.The value is an integer. The options are as follows:0: No1: YesThe default value is 0.
164Indicates whether an AR proactively releases a call upon the receiving of an ACK message carrying media information after media negotiation is completed using an INVITE message and a 200 OK message on a SIP trunk.The value is an integer. The options are as follows:0: Yes1: NoThe default value is 0.
166 to 199Reserved.
200Switch for generating failure-specific CDRs.Enumerated value. The options are as follows:0: No CDRs are generated for any failures.1: CDRs for all failures are generated.2: CDR for trunk failures are generated.The default value is 0.
201Switch for changing a called number in the CDR.Enumerated value. The options are as follows:0: Before change1: After changeThe default value is 0.
202Indicates whether the narrowband trunk plays tones on the narrowband trunk.Enumerated value:0: No1: YesThe default value is 0.
203Switch for direct-dialing through Virtual User (VU) numbers.Enumerated value. The options are as follows:0: Disable1: EnableThe default value is 0.
204Switch for displaying the original caller number.Enumerated value. The options are as follows:0: Not display the original caller number1: Display the original caller numberThe default value is 0.
205Specifies an interval for processing a new registration request upon a SIPUE authentication failure.The value is an integer that ranges from 0 to 300, in seconds.The default value is 10.
206Switch for generating local-survival bill when deactivate.Enumerated value. The options are as follows:0: Not generate.1: Generate.The default value is 0.
207Switch for generating system bill.Enumerated value. The options are as follows:0: Not generate.1: Generate.The default value is 1.
208Switch for checking voice password.Enumerated value. The options are as follows:0: Not check1: CheckThe default value is 1.NOTE:To secure devices, the password complexity verification function must be enabled.
209Specifies whether to send the number when ISDN trunk group reaches the minimum number length.Enumerated value. The options are as follows:0: Not send1: SendThe default value is 1.
210Switch for enabling preempted registration.Enumerated value. The options are as follows:0: Allowed1: Allowed when not busy2: All forbiddenThe default value is 1.
211Whether to transparently transfer the number sign (#) in the called number for a trunk-based outgoing call.Enumerated value. The options are as follows:0: The number sign (#) is not transparently transferred.1: The number sign (#) is transparently transferred.The default value is 0.
212Path refresh timer length.The value is an integer that ranges from 0 to 5, in seconds.The default value is 2.
213Change sdp port when re-invite negotiation.Enumerated value. The options are as follows:0: Yes1: NoThe default value is 0.
214Whether restrict outgoing right by caller.Enumerated value. The options are as follows:0: No1: YesThe default value is 1.
215Caller release call duration of not offhook under called control mode.The value is an integer that ranges from 60 to 65535, in seconds.The default value is 180.
216Caller detects dtmf send high frequency tone duration of offhook under called control mode.The value is an integer that ranges from 10 to 65535, in seconds.The default value is 30.
217Send high frequency tone length under called control mode.The value is an integer that ranges from 1 to 65535, in seconds.The default value is 2.
218Whether to set single Pre-dial number that can trigger outgoing dial-tone.The value is an integer that ranges from 0 to 0xFFFF.The default value is 0xFFFF (not provided).
219Whether UAC responds to 200OK when the refresh is initiated by UAC and UAS sends the update message.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
220Specifies whether to add cdsc in sdp when support t38 fax.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
221Specifies whether pra trunk play ringback tone to peer.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
222Specifies whether to check country and area code when a trunk calls out.Enumerated value. The options are as follows:0: No1: YesThe default value is 1.
223Specifies whether to play the voice tone when abnormal call.Enumerated value. The options are as follows:0: Yes1: NoThe default value is 0.
224Specifies whether through payload of voice when call from IP to IPEnumerated value. The options are as follows:0: No1: YesThe default value is 1.
225Specifies whether to send 403 or 486 message when no circuit/channel available.Enumerated value. The options are as follows:0: 4031: 486The default value is 0.
226Specifies whether to enable DDoS attack prevention.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
227Indicates the threshold of DDoS attack prevention.The value is an integer that ranges from 0 to 1000, in packets per second.The default value is 20.
228Specifies whether to refresh timer when recv alerting again.Enumerated value. The options are as follows:0: Not refresh1: RefreshThe default value is 1.
229Determines whether the called party payment service is supported.Enumerated type. The options are as follows:0: The called party payment service is not supported.1: The called party payment service is supported.The default value is 1.
230Determines whether the callingnumber field is hidden in the Setup message of Dss1.Enumerated type. The options are as follows:0: The callingnumber field is not hidden in the Setup message of Dss1.1: The callingnumber field is hidden in the Setup message of Dss1.The default value is 0.
231Determines whether the R2 trunk sends the A4 message if the interim dialing timer times out on the called party side.Enumerated type. The options are as follows:0: The R2 trunk does not send the A4 message if the interim dialing timer times out on the called party side.1: The R2 trunk sends the A4 message if the interim dialing timer times out on the called party side.The default value is 0.
232Determines whether the DSS1 trunk listens to the ringback tones of the remote device.Enumerated type. The options are as follows:0: The DSS1 trunk does not listen to the ringback tones of the remote device.1: The DSS1 trunk listens to the ringback tones of the remote device.The default value is 0.
233Determines whether the eri-location field is in the cause field of the SIP message.Enumerated type. The values range from 0 to 1.0: The eri-location field is not in the cause field of the SIP message.1: The eri-location field is in the cause field of the SIP message.The default value is 1.
234Determines the negotiation mode of the fax in a multi-M-line scenario.Enumerated type. The options are as follows:0: T38 is used.1: The transparent transmission is used.2: The order is used.The default value is 2.
235Specifies whether to convert the G.729 to G.729B when an H323 trunk is used.Enumerated type. The options are as follows:0: No1: YesThe default value is 0.
236Indicates whether the AR or the IMS registered with the AR controls the three-party call service.Enumerated type. The options are as follows:0: AR1: IMSThe default value is 0.
237Indicates whether the AR or the IMS registered with the AR controls the call forwarding service.Enumerated type. The options are as follows:0: AR1: IMSThe default value is 0.
238Indicates whether the AR plays call hold tones to the remote end when a user presses the hookflash.Enumerated type. The options are as follows:0: No1: YesThe default value is 1.
239Indicates the SIP message caching timeout.The value is an integer that ranges from 0 to 2000, in milliseconds.The default value is 1000.
240Reserved.
241Reserved.
242Specifies whether to enable the simplified three-party call service and call wait flow.The value can be 0 or 1.0: Disable1: EnableThe default value is 0.
243Specifies whether to enable an FXO interface to work in auto-answer mode.Enumerated value. The options are as follows:0: Disable1: EnableThe default value is 1.
244Specifies a no answering timer for an FXO interface when the FXO interface works in non-auto answering mode.The value is an integer that ranges from 5 to 10, in seconds.The default value is 5.NOTE:This configuration takes effect only after pbx number-parameter 243 0 is configured.
246Specifies whether to enable polarity reversal on an FXS interface.Enumerated value. The options are as follows:0: Disable1: Enable reversal on seizure (ROS)2: Enable reversal on answer (ROA)NOTE:ROS and ROA cannot be enabled simultaneously.The default value is 0.
247Indicates whether the R2 trunk supports accounting pulse in the Argentina and Brazil templates.Enumerated value. The options are as follows:0: No1: YesThe default value is 0.
248Specifies whether to enable automatic IVR two-stage dialing.Enumerated value. The options are as follows:0: Disable1: EnableThe default value is 0.NOTE:Automatic IVR two-stage dialing requires short codes with 0 as the dialing index. If automatic IVR two-stage dialing is enabled, the system automatically forwards a call to a number with 0 as the dialing index after playing greetings.When automatic IVR two-stage dialing is enabled, do not dial a number while the system is playing greetings. Otherwise, the dialing takes no effect.Automatic IVR two-stage dialing does not support the time section index.
249Indicates whether to replace the display header left empty in the H323 Q.931 message on the local device with the display header in the Q.931 message sent for H323 incoming calls.The value is an integer. The options are as follows:0: No1: YesThe default value is 1.
Asterisk & FreePBX: How To Create AMI User?

Asterisk & FreePBX: How To Create AMI User?

AMI (Asterisk Management Interface) is used for third party applications to work properly with Asterisk (FreePBX). Applications can retrieve updated information about system events via AMI and send some commands to Asterisk.

To use AMI on Asterisk and FreePBX, you need to define a user and password. You can create this in two ways:

  1. By editing the /etc/asterisk/manager.conf file
  2. By creating a user with Asterisk Manager in FreePBX.

Let’s examine both methods in more detail.

Creating an AMI User on Asterisk:

  1. Connect to the Asterisk server via SSH.
  2. Open the /etc/asterisk/manager.conf file with an editor.
nano /etc/asterisk/manager.conf
  1. Enter the AMI settings in the [general] section:
[general] 
enabled = yes 
port = 5038 
bindaddr = 0.0.0.0 
displayconnects = no
  1. Create a new user under the [general] section:
[amiuser] 
secret = 123456 
deny = 0.0.0.0/0.0.0.0 
permit = 192.168.0.0/255.255.255.0 
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message 
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message 
writetimeout = 5000

Note: In this example, the username is amiuser and the password is 123456. It only accepts requests from the 192.168.0.0/24 network, but all access is read and write.

  1. Save and close the file.
  2. Reinstall the Asterisk Manager module:
asterisk -rx "manager reload"

Creating AMI User on FreePBX:

  1. Click Settings in the top navigation bar and then Asterisk Manager Users:
Asterisk Manager Users Menu

2. Click the Add Manager button to add a new user.

FreePBX Add Manager

3. Here you can see the settings window for a new user. Complete the relevant fields here:

AMI User Properties
  • Manager Name: Enter the AMI Username (no spaces).
  • Manager Secret: Enter the password for the AMI User.
  • Deny: Here you can specify the IP address and subnet mask that you want to restrict connection with AMI. You can use the “&” symbol to specify multiple networks, eg 192.168.1.0/255.255.255.0&192.168.2.0/255.255.255.0
  • Permit: Specify the network or unique IP address to which you allow AMI connection.
  • Write Timeout: Enter the timeout period for the execution of the AMI command.
  1. Set access permissions using the Permissions tab.
AMI Permissions

5. Click the Submit button at the bottom right to save the settings and then click Apply Config appears at the top left to complete the process.

openSIPS Installation Steps

openSIPS Installation Steps

openSIPS is a multi-purpose SIP server that is used by many telephony service providers and offers Class 4, Class 5, wholesale VoIP, enterprise PBX, virtual PBX, SBC, load balancing IMS platforms, call centers features and more. In this article, you can find the installation steps of openSIPS on Debian 10.

openSIPS is a high performance SIP server running on Linux that needs very little resources. Therefore, many telecom operators develop solutions with openSIPS. If you want to use openSIPS in your VoIP applications, you can follow the installation instructions below.

openSIPS Installation Steps

1. Components Used in openSIPS Installation & Versions:

  • Debian v10 (Buster) x64 minimal install (netinst)
  • OpenSips v3.0
  • OpenSips GUI v8.3.0
  • Apache v2.4
  • PHP v7.3
  • MariaDB v10

2. Pre Installation Tasks

To install openSIPS, you will first need a fresh Debian installation. You can download and install the amd64 netinst CD image from this link. Debian is very easy to install, you can also install it by following this video that I prepared.

After installing Debian, complete the installation of the following packages:

apt update && apt upgrade -y && apt -y install m4 git nano sudo curl dbus apache2 lsb-release

Normally, you can install the “monit” package as an option for monitoring, but at the time I wrote the article, it was removed from debian repos due to some vulnerabilities on the monit package. In case the situation changes, you can find the related setup command below:

apt -y install monit

3. PHP Installation

First install dependencies:

apt -y install curl apt-transport-https ca-certificates

Add PHP repo:

wget -O /etc/apt/trusted.gpg.d/php.gpg https://packages.sury.org/php/apt.gpg echo "deb https://packages.sury.org/php/ $(lsb_release -sc) main" > \ /etc/apt/sources.list.d/php.list 

After that, install PHP packages:

apt update && apt -y install php7.3 php7.3-gd php7.3-mysql php7.3-xmlrpc php-pear php7.3-cli php-apcu php7.3-curl php7.3-xml libapache2-mod-php7.3 

Install PHP MDB2 library with pear:

pear install MDB2#mysql

Edit PHP.ini file and change short_open_tag variable to On:

sed -i "s#short_open_tag = Off#short_open_tag = On#g" /etc/php/7.3/apache2/php.ini

4. MariaDB Installation

Get gpg keys needed for MariaDB repo and install necessary packages:

apt-key adv --recv-keys --keyserver hkp://keyserver.ubuntu.com:80 0xF1656F24C74CD1D8 curl -sS https://downloads.mariadb.com/MariaDB/mariadb_repo_setup | sudo bash apt update && apt -y install mariadb-server 

After that edit my.cnf file as below:

nano /etc/mysql/my.cnf

To disable Strict mode and use default openSIPS latin1 character set, add these lines under [mysqld] header:

sql_mode='' 
character-set-server = latin1 

Restart MariaDB service:

systemctl restart mariadb

5. openSIPS Installation

Add gpg key:

apt -y install dirmngr && apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B

Add openSIPS repos:

echo "deb https://apt.opensips.org $(lsb_release -sc) 3.0-releases" >/etc/apt/sources.list.d/opensips.list 
echo "deb https://apt.opensips.org $(lsb_release -sc) cli-nightly" >/etc/apt/sources.list.d/opensips-cli.list 

Install openSIPS packages:

apt update && apt -y install opensips opensips-cli opensips-*-module opensips-*-modules python3-mysqldb python3-sqlalchemy python3-sqlalchemy-utils 

6. Database Installation

Create opensips user on MariaDB and grant rights:

mysql -p 
> 
CREATE USER 'opensips'@'localhost' IDENTIFIED BY 'opensipsrw'; 
GRANT ALL PRIVILEGES ON opensips.* TO 'opensips'@'localhost'; 
FLUSH PRIVILEGES; 
exit 

Run database installation script:

opensips-cli -x database create 

The script will ask you the database URL. Enter mysql://opensips:opensipsrw@localhost and choose default (install all tables).

7. Generating Configuration File

Run configuration generator script to generate configuration file:

/usr/sbin/osipsconfig 

Choose GenerateOpenSIPS Script > Residential Script > Configure Residential Script. Choose all items other than TLS by using space bar. Use Q to go to previous menu and schoose Generate Residential Script. Script will generate a configuration file and will promt the file name on screen. Replace opensips.cfg file with the generated one and give necessary rights:

mv /etc/opensips/opensips.cfg /etc/opensips/opensips.cfg.orig 
cp /etc/opensips/[üretilen konfig dosyası] /etc/opensips/opensips.cfg 
chmod 644 /etc/opensips/opensips.cfg 

8. Additional Configurations:

Write server IP address in opensips.cfg file:

nano /etc/opensips/opensips.cfg

write server IP addresses in two lines starting with listen= :

listen=udp:192.168.0.1:5060
listen=tcp:192.168.0.1:5060

Then check if the configuration file is valid or not:

opensips -C /etc/opensips 

If there is an error in the file, it will appear on the screen. Correct the errors, otherwise run the opensips service with the new configuration file by using the following command:

systemctl restart opensips 

9. GUI Installation

Download openSIPS GUI via git:

git clone -b 8.3.0 https://github.com/OpenSIPS/opensips-cp.git /var/www/opensips-cp

Create openSIPS GUI table on database:

cd /var/www/opensips-cp/config 
mysql -p opensips < db_schema.mysql 

10. Regular Collection of Statistics

Add the necessary script into cron.d folder and restart cron service:

cd /var/www/opensips-cp/config
cp tools/system/smonitor/opensips_stats_cron /etc/cron.d/
systemctl restart cron

11. Monit Configuration (Optional)

Add the necessary line into monitrc file and restart monit service:

echo -e "set httpd port 2812 and\nallow localhost" >> /etc/monit/monitrc
systemctl restart monit

12. Global Configurations

Open GUI config PHP file and edit as follows:

nano +30 /var/www/opensips-cp/config/boxes.global.inc.php 
// monit host:port 
$boxes[$box_id]['monit']['conn']="127.0.0.1:2812"; 
$boxes[$box_id]['monit']['user']="admin"; 
$boxes[$box_id]['monit']['pass']="admin"; 
$boxes[$box_id]['monit']['has_ssl']=0;

13. Apache Configuration

Define Virtual Hosts on Apache by using the commands below:

cat >> /etc/apache2/sites-available/opensips.conf << EOF 
<VirtualHost *:80> 

DocumentRoot /var/www/opensips-cp 

<Directory /var/www/opensips-cp/web>
     Options Indexes FollowSymLinks MultiViews
     AllowOverride None
     Require all granted 
</Directory> 

<Directory /var/www/opensips-cp>
     Options Indexes FollowSymLinks MultiViews
     AllowOverride None
     Require all denied 
</Directory> 
Alias /cp /var/www/opensips-cp/web 

<DirectoryMatch "/var/www/opensips-cp/web/tools/.*/.*/(template|custom_actions|lib)/">
      Require all denied 
</DirectoryMatch> 

</VirtualHost> 

EOF 

Disable default site, enable openSIPS GUI site, change owner of the folder and restart Apache:

a2dissite 000-default 
a2ensite opensips 
chown -R www-data. /var/www/opensips-cp 
systemctl restart apache2 

Finally the installation is finished. Use http://ipadress/cp URL with admin / opensips credentials to access openSIPS GUI.

What is ENUM? ENUM Syntax

What is ENUM? ENUM Syntax

ENUM (Telephone Number Mapping, E.164 Number to URI Mapping) is an addressing protocol that converts telephone numbers to URI format (name@domain). This allows you to access a SIP, H.323 or other Internet phone user by dialing a phone number.

The ENUM function aims to ensure that users can be accessed anywhere in the world with the same number, best quality and the cheapest way. ENUM maps a phone number to an Internet address in the DNS system. Thus, a user with an ENUM number can broadcast the DNS record to which the call will be routed. Even different routes can be defined for different types of calls (fax, video, etc.).

It is possible to obtain an ENUM record as if it were a domain name. Nowadays, you can obtain this registration free of charge through many registration services and VoIP service providers.

ENUM Syntax

ENUM allows normal phone (E.164) numbers to be displayed as DNS names ending in e164.arpa. A number can be decoded for one or more predefined services.

For example, a telephone number + 90-312-555-1234 will be displayed as 4.3.2.1.5.5.5.2.1.3.0.9.e164.arpa after issuing rules defined in RFC 3761 and below:

  1. All characters except digits are removed. (“+90-312-555-1234” becomes “903125551234″)
  2. A period (“.”) Is placed between each number. (“9.0.3.1.2.5.5.5.1.2.3.4”)
  3. The order of the numbers is reversed. (“4.3.2.1.5.5.5.2.1.3.0.9”)
  4. .e164.arpa is added to the end of the array. (“4.3.2.1.5.5.5.2.1.3.0.9.arp a”)

To respond to this syntax, the DNS server must have a record that looks like this:

$ ORIGIN 4.3.2.1.5.5.5.2.1.3.0.9.barley.
   NAPTR 10 100 "u" "E2U + sip" "! ^. * $! Sip: fatih.erikci@fatiherikci.com!" .
   NAPTR 10 101 & quot; u & quot; & quot; E2U + h323 & quot; .
   NAPTR 10 102 "u" "E2U + msg" "! ^. * $! Mailto: fatih.erikci@fatiherikci.com!" .

In this record you see three different routing sequences for the address 4.3.2.1.5.5.5.2.1.3.0.9. The first is SIP, the second is H.323 and the third is the SMTP response. Device selects which service to communicate by using these records.

How It Works?

The operating principle of ENUM is similar to the DNS queries we use on the Internet. DNS NAPTR resource records are used in queries.

ENUM Query and Call
  1. The phone calls an E.164 number (+90-312-555-1234)
  2. Gateway translates it (4.3.2.1.5.5.5.2.1.3.0.9.e164.arpa) and asks the DNS server.
  3. The DNS server responds to this query with a URI (sip: fatih.erikci@fatiherikci.com).
  4. The gateway sends the call to the SIP server as a SIP URI call.
  5. The SIP server rings the IP phone registered with the URI.
How To Save Energy With Cisco IP Phone?

How To Save Energy With Cisco IP Phone?

If you are using Cisco solutions in your phone and network infrastructure, there are several methods that can help you to save energy and reduce your electricity costs. In this article you can find how to save energy in your IP phone & network infrastructure.

As you know, unlike analog phones, IP phones require a power supply to work. They can get their power even from power adapters or network devices that can give power through PoE.

What is PoE?

PoE is an abbreviation for Power over Ethernet. In a very simple sense, PoE is a protocol which is defined by EEE 802.3af (15.4W max.) And 802.3at (30W max.) standards and it enables devices (IP phones, wireless APs. etc.) to get their power through a network cable.

Since IP telephones do not need much energy, they can easily work with the 802.3af standard. In this standard, devices are classified according to their power requirements. You can find these classes and power requirements in the following table:

802.3af ClassMax. Power (W)
315.4
27
14
015.4

The following table shows typical and maximum power consumption of current Cisco IP phones:

Model802.3af ClassIdle (W)Maximum (W)
781112.63.8
782112.63.8
784112.63.8
786112.63.8
881123.96.5
884123.96.5
884523.96.4
885143.99.8
886144.214.2
886544.214.7

What About Savings With IP Phone?

Cisco IP phones that work with CUCM have two different energy saving modes, Power Save and Power Save Plus. Now let’s examine them in detail:

IP Phone – Power Save Mode

In this mode, the display’s backlight does not light when the phone is not in use and the phone remains registered with the CUCM, can receive incoming calls, and also users can make calls. The CUCM phone configuration menu (CUCM Administration > Device > Phone > related phone) has special configuration options to manage background lighting on certain days, at a specified time interval, and when the phone is not used for a certain period of time. You can find these configuration options and descriptions below:

OptionExplanation
Days Display Not ActiveSpecifies the days which the screen is not turned on. You can select the day which the screen is off from the list. Hold down the Ctrl key for multiple selections.
Display On Time Specifies the time at which the display will be turned on. This field should be written in 24-hour format. For example, if you want it to open at 8:30 in the morning, you should type 08:30. If this field is left blank, the screen will turn on at 0:00.
Display On Duration Specifies the length of time the display stays on. For example, if you enter 08:00, the phone display will remain on until 16:30 with the configuration above. If no information is entered in this field, the display will remain on until midnight.
Display Idle TimeoutSpecifies how long after which the display turns off when the phone is not in use. If you want it to close after 1 hour idle, just enter 01:00.

Note 1: In Power Save mode, the phone exits the power saving mode when the user lifts the handset or press any button when the phone’s display or backlight is off.

Note 2: 7811 series phones do not support energy saving in Power Save mode because they do not have backlight.

As you can see in the second table, there is about 30% difference between the idle and max amount of power they draw. Therefore, using Power Save mode can save approximately 20% energy with a very rough calculation. You also do not need extra hardware software or licenses to implement this mode.

Power Save Plus Mode (EnergyWise)

In the Power Save Plus mode, the phones go to sleep in the specified time or day, as in power save mode. In this case, incoming calls cannot be received because the phones do not register to the CUCM, and users must press the Select key and wake the IP phone to make calls. The advantage is that the energy consumption in sleep state is 1W.

Note: Power Save Plus mode is offered as part of a solution called Cisco Energy Management Suite and needs to be supported by switches on your network. Your CUCM version needs to be 8.6 and higher, and the firmware version of the phones needs to be 9. (2) 1 and higher.

Likewise, the CUCM phone configuration menu for Power Save Plus configuration has special configuration options to manage the sleep mode on specific days, at a specified time interval, and when the phone is not used for a certain period of time. You can find these configurations and descriptions below:

OptionExplanation
Enable Power Save Plus Sets the days on which this feature will be activated. You can select multiple days by holding down the Ctrl key.
Phone On Time Specifies the time at which the phone is switched on. If you want it to open at 7:30 in the morning, you must enter 07:30.
Phone Off Time Sets the time at which the phone goes to sleep. If you want to go to sleep at 6:00 in the evening, you must enter 18:00.
Phone Off Idle Timeout Determines how long it takes for the phone to go to sleep inactive. This feature is active in the following situations:
If the phone is woken up during Off Time (with the select key).
If the phone is awakened by the switch.
If the time has come off but the phone has continued to be used.
Enable Audible Alert When activated, the phone emits an audible tone (ringing tone) 10 minutes before it goes into sleep mode.
EnergyWise Domain The EnergWise domain name where the phone is located.
EnergyWise Secret The password that the phone uses to talk to devices in the EnergyWise domain.
Allow EnergyWise Overrides Determines whether policies printed from the EnergyWise controller have priority over the CUCM configuration.

Last Words

In terms of energy saving, Pover Save and Power Save Plus modes provide a great advantage for organizations which has Cisco IP phone infrastructure, and they can easily be applied. In this way, you can reduce operational costs by increasing energy efficiency.

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