Release note VOIPv8.12 - addpac.su · APOS Release Note Voice over IP Added Features 1. STUN STUN (Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators

APOS Release Note Voice over IP

APOS Release Note Release 8.23 Feb. 2006

Feb. 2006

Technical Laboratory AddPac Technology Co., Ltd.

AddPac Technology Proprietary & Documentation 95 - 1


[ Table of Contents ]

Added Features..................................................................................... 6

1. STUN ..........................................................................................................................6

Network Configuration ........................................................................................................................ 7

Related Commands.............................................................................................................................. 8

2. udp-checksum...........................................................................................................9


Related Commands............................................................................................................................ 10

Default: Disable...................................................................................................................10

3. qos-threshold ............................................................................................................2



4. PPTP Route Data .......................................................................................................4



5. Connection Delay......................................................................................................8



6. caller-id gain ............................................................................................................10

Network Configuration ...................................................................................................................... 10

Related Commands.............................................................................................................................11

7. Timeout tvcc ............................................................................................................12



8. Timeout tttl fixed .....................................................................................................14



Default: Disable...................................................................................................................15

9. FXS Port PSTN-Backup-Port ....................................................................................2



10. Restricting Call Duration ..........................................................................................5



11. Changing FAX Scenario ...........................................................................................7


12. busyout monitor callagent .......................................................................................8





13. rtp-nat-pat ................................................................................................................10


Related Commands.............................................................................................................................11

14. SIP Call-transfer-mode [basic/attended] ...............................................................12



15. SIP Conference-server............................................................................................16



16. SIP enable-ping .......................................................................................................19



17. SIP Media-channel [early|late]................................................................................22



18. SIP Remote-party-id ................................................................................................25



19. SIP route-by-auxiliary .............................................................................................28


20. SIP Call-Diversion ...................................................................................................29



21. SIP user-config call-diversion................................................................................31



22. SIP set-local-domain...............................................................................................33


23. SIP set-local-host ....................................................................................................35


24. SIP SRV enable........................................................................................................37



25. SIP rel1xx .................................................................................................................39



26. SIP Response ..........................................................................................................41





27. SIP dtmf-relay h245-signal .....................................................................................43



Default: Disable...................................................................................................................44

28. Secondary MGC Registration ................................................................................45



29. Separate MGCP Registration .................................................................................47



30. MGCP busyout-timer...............................................................................................49



31. MGCP dtmf-relay dual-mode ..................................................................................51



32. MGCP epid-type.......................................................................................................53



33. FAX protocol ‘multi-session-t38’ ...........................................................................55



34. MGCP force-local-rt.................................................................................................57



35. E1/T1 compand-type [au-law|ua-law] ....................................................................59


36. E1/T1 isdn overlap-sending ...................................................................................60



37. E1/T1 clock slave-main...........................................................................................62


38. ISDN PRI numbering-type ......................................................................................63



39. ISDN PRI numbering-plan ......................................................................................65





40. ISDN immediate-disc ..............................................................................................67



41. R2 dtmf gain Change ..............................................................................................69



Removed Feature ................................................................................ 71

42. Announcement ........................................................................................................71

Model: AP1005Troubleshooting ........................................................................................71

Troubleshooting..................................................................................................................72

43. Command Error, “clear-down-tone”......................................................................72

44. Async Modem..........................................................................................................72

45. Voip-interface...........................................................................................................72

46. Call Forwarding Errors When Ring Timer Expires...............................................72

47. PAT Table Display....................................................................................................72

48. RTP Payload Timestamp Value Errors ..................................................................72

49. STUN Disabled When IP Address is Changed......................................................72

50. An INVITE Message Sent Before SIP Registration Through AP160 ...................72



Added Features 1. STUN

STUN (Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)) supported by APOS™ is designed based on the features of RFC3489. STUN is a standard protocol based on UDP, and allows AddPac Gateway to connect to another gateway that has a public IP in an NAT/firewall environment. The “public-ip” command enables a connection to a public IP in a private environment. If the public IP mapped with a private IP is changed frequently, the manager should set the “public-ip” command again. The STUN feature of APOS can resolve this issue. If the interface IP of a device is changed, this feature allows you to send an STUN message to the STUN server and receive the changed public IP to update the public IP. If a public IP is changed from the NAT server, you can check the changed public IP by checking periodical STUN messages. The STUN server operates just like other servers. The STUN server has a simple request and response architecture. The server, which is located at a public IP bandwidth, processes the data received from STN Client and sends the result to STUN Client. Number of the standard UDP port used by STUN is 3478.

Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.



Network Configuration

WAN(IP Network)

AddPacVoIP Gateway

Analog Phone(1000)

Stun Server Gatekeeper

AddPacRouter (NAT-Server)

Binding RequestBinding Response

RRQRCF

Binding Request

RRQRCF

Public IP 192.168.1.2

Binding RequestBinding Response

Retry Timer

Private IP Address: 10.1.1.1Public IP Pool: 192.168.1.2 ~ 192.168.1.10

1000 Alias 192.168.1.2

Public IP 192.168.1.3

Local IP Changed

IP Address 192.168.2.200IP Address192.168.50.100

IP Address 10.1.1.2

Binding Response

1000 Alias 192.168.1.3

[Figure 1] VoIP Gateway STUN Feature



Related Commands

Enabling STUN Feature

Step Command Description 1 # config Switches to APOS command setting mode.

2 (config)# interface ethernet 0.0 Switches to interface setting mode.

3 (config-ether0.0)# ip stun-server 61.33.161.111 Sets the IP address of the STUN server.

4 (config-ether0.0)# ip stun-server retry-time 10 Sets the interval of sending an STUN

request message periodically.

Disabling STUN Feature

Step Command Description 1 (config-ether0.0)# no ip stun-server Disables the STUN feature.



2. udp-checksum

In the past, some sensitive terminals that check UDP checksum have not been able to receive RTP properly while making a VoIP call. If UDP checksum is enabled by using the relevant feature, even a terminal that checks UDP checksum could receive RTP properly.



[Figure 2] VoIP Gateway udp-checksum Enable



Related Commands

Enabling udp-checksum


2 (config)# ip udp-checksum enable Enables udp-checksum.

Disabling udp-checksum

Step Command Description 1 (config)# no ip udp-checksum Initializes this feature.

Default: Disable



3. qos-threshold

The APOS VoIP product family allows you to check data on delay, jitter, and packet loss of the recent calls. If you set the threshold value of delay, jitter, and packet loss that may affect calling quality, the data on a call that exceeds the threshold value will be transferred to the SNMP trap. Also, the APOS VoIP product family provides the data on quality of recent calls when a request for SNMP Get is made. You can check the data from SNMP MIB. Currently, SIP supports the feature.



WAN(IP Network)

AddPacVoIP Gateway-A

Analog Phone

voice service voip:qos-threshold delay 500qos-threshold jitter 300qos-threshold packet-loss 100

GET SNMPv2-MIB

AddPacVoIP Gateway-B

Analog Phone

RESPONSE SNMPv2-MIB

SNMP Management

[Figure 3] Sending TOS Information When a Request for SNMP Get is Made



Related Commands

Enabling QOS-Threshold

Step Command Description 1 # config Switches to APOS command setting

mode.

2 (config)# voice service voip Switches to interface setting mode.

3 (config-vservice-voip)# qos-threshold delay 1000

(config-vservice-voip)# qos-threshold jitter 500

(config-vservice-voip)# qos-threshold packet-loss 5

Sets qos-threshold.

Initializing QOS-Threshold

Step Command Description 1 (config-vservice-voip)# no qos-threshold delay

(config-vservice-voip)# no qos-threshold jitter

(config-vservice-voip)# no qos-threshold packet-loss

Initializes qos-threshold.



4. PPTP Route Data

Point-to-Point Tunneling Protocol (PPTP) supported by APOS™ conforms to RFC2637. If you can access the PPTP server through the LAN interface, you could use the PPTP feature to configure Virtual Private Network (VPN). PPTP supported by AddPac Gateway is a client feature that allows you to access the PPTP server; thus, a PPTP server must exist on the Internet. In the past, VoIP and data have been transferred through a tunnel by using the traditional PPTP route tunnel; however, you can execute the newly added “PPTP route data” command to enable only VoIP to be transmitted through a tunnel. Perform the setting below to configure a tunnel through which VoIP packets will be forwarded and to forward data packets to WAN: (config-ether0.0)# encapsulation ppp-pptp For information on the detailed configuration procedure, refer to ‘Related Commands’. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.




[Figure 4] VoIP Gateway PPTP



Related Commands

Enabling PPTP Route Data Feature


2 (config)# interface ether 0.0 Switches to interface setting mode.

3 (config-ether0.0)# no ip address Does not set an IP address.

4 (config-ether0.0)# encapsulation ppp-pptp Sets the network protocol to PPTP. (Note:

Only if the encapsulation ppp-pptp is

enabled, interface pptp 0 will be created.)

5 (config-ether0.0)# pptp ip remote IP-ADDRESS Sets the IP address of the PPTP server.

6 (config-ether0.0)# pptp route data Transfers data to the interface PPTP.

7 (config-ether0.0)# ppp authentication chap callin Sets the PPP authentication method to

Chap. (If you want to set the PPP

authentication method to PAP, refer to the

Quick Operation Guide.)

8 (config-ether0.0)# ppp chap hostname WORD Sets the user ID of Chap to “addpac”.

9 (config-ether0.0)# ppp chap password LINE Sets the password of Chap to “1234”.

10 (config-ether0.0)# no ppp ipcp ms-dns Disables the setting that allows you to

receive the IP address of DNS from the PPP

server.

11 (config-ether0.0)# no ppp ipcp default-route Disables the setting that allows you to

receive the IP address of the default

router from the PPP server (Important).

12 (config-ether0.0)# exit Disables the mode for Ethernet Interface 0.0.

13 (config)# interface pptp0 Switches to interface pptp 0 setting mode.

14 (config-pptp0)# ip address IP-ADDRESS SUBNET-

MASK

Sets an IP address. (For information on how

to set DHCP and PPPoE, refer to the Quick

Operation Guide.)

15 (config-ether0.0)# exit Disables the mode for Ethernet Interface 0.0.

16 (config)# route 0.0.0.0 0.0.0.0 ROUTER-IP Sets the default router.

17 (config)# ip-policy ip host voip-interface any route-

if ether0.0

Allows you to transfer data to a public

network and to send VoIP traffic to a private

network.

18 (config)# exit Disables setting mode.



Disabling PPTP Feature

Step Command Description 1 (config-ether0.0)# no encapsulation ppp-pptp Disables the PPTP feature.

Note: MS-Chap, which is one of the PPP authentication methods, is not supported. Default: Disable



5. Connection Delay

Typically, PBX does not deliver a message related to connections; thus, call duration of the actual users may be different from the amount of time it will take to perform billing by using a CONNECT message if a network where PBX is used is configured. In such a case, you can enable a CONNECT message to be forwarded in a specific amount of time by executing the relevant commands. Note that an outgoing call should be connected by one stage dialing. The example below shows that “prefix” configuration is performed on the VoIP gateway for “one stage dialing”.



[Figure 5] Sending VoIP Gateway Connect Message



Related Commands

Enabling delayed-connect Feature

Step Command Description 1 #

# config

Switches to APOS command setting

mode.

2 (config)# voice service voip Starts the VoIP service.

3 (config-vservice-voip)# delayed-connect <1 - 254> Enables the delayed-connect feature.

Disabling delayed-connect Feature

Step Command Description 1 (config-vservice-voip)# no delayed-connect Disables the delayed-connect feature.

Default: Disable



6. caller-id gain

AddPac Gateway receives the CID information as well when receiving a VoIP call, and displays the information on the phone. Typically, frequency modulation is performed on a network, while frequency demodulation is performed at the CID terminal. Frequency modulation should meet the standard characteristics of 1,200 Baud defined under the Bellcore recommendations for a channel for omni-directional data transfer. If AddPac Gateway delivers CID properly but the other party cannot recognize the CID, you can change the signal level required for modulation by using the relevant commands. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 6] Changing CID Gain of VoIP Gateway



Related Commands

Changing caller-id gain Value


# config


mode.

2 (config)# voice-port 0/0 Starts voice port settings.

3 (config-voice-port-0/0)# caller-id gain <-18 ~ -6> Changes the caller-id gain value.

Initializing caller-id gain Value

Step Command Description 1 (config-voice-port-0/0)# caller-id gain -13 Initializes the caller-id gain value.



7. Timeout tvcc

If voice is transmitted and a connection cannot be made in a configuration where a VoIP call is made through PBX even if the recipient hangs up the phone, you can enable a Disconnect message to be sent to the caller in a specific amount of time by executing the related commands. If a Connect message is not sent from PBX (For instance, an internal user does not answer a call), you can enable a Disconnect message to be sent to the caller in a specific amount of time set in the timer by executing the relevant commands. Note that an outgoing call should be connected by one stage dialing just like in Connection Delay. Also, the command below must be set to enable voice-confirmed-connect timeout: (config-vservice-voip)# voice-confirmed-connect



[Figure 7] Sending VoIP Gateway Disconnect Message



Related Commands

Enabling voice-confirmed-connect timeout Feature


# config


mode.

2 (config)# voice-port 0/0 Starts the voice-port setting.

3 (config-voice-port-0/0)# timeout tvcc <0 - 1800> Enables the voice-confirmed-connect

timeout feature.

Disabling voice-confirmed-connect timeout Feature

Step Command Description 1 (config-voice-port-0/0)# no timeout tvcc Disables the voice-confirmed-connect

timeout feature.

Default: Disable



8. Timeout tttl fixed

In a configuration environment where a VoIP call is made through a gatekeeper, the gateway should be operated based on the time to live (ttl) timer setting. If you want to configure an environment where AddPac VoIP Gateway inter-works with a gatekeeper based on your own ttl timer setting, execute the related commands. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


WAN(IP Network)

AddPacVoIP Gateway

Analog PhoneGateKeeper

RRQRCF

RRQRCF

voice-service-voipTimeout tttl <10-86400> fixed

ttl timeoutGatekeeperTTL timeout

[Figure 8] VoIP Gateway TTL Timeout



Related Commands

Enabling ttl timer fixed Feature


# config


mode.

2 (config)# voice service voip Enables the VoIP Service features.

3 (config-vservice-voip)# timeout tttl <10 - 86400> fixed Sets the ttl timer.

Disabling ttl timer fixed Feature

Step Command Description 1 (config-vservice-voip)# no timeout tttl Initializes the ttl timer.

Default: Disable



9. FXS Port PSTN-Backup-Port

If the gateway is in busyout mode, the cause would be one of the following: First, the power supply is blocked. Second, the LAN interface of the VoIP gateway is down. Third, the VoIP gateway cannot be connected to the other VoIP gateway because the gatekeeper, MGC, and proxy server are down. The SIP proxy server does not control the busyout mode. If a VoIP-peer connection fails, enable the hunt feature to perform hunting for the POTS-peer setup at the PSTN backup port. If the gateway is in busyout mode, it will be difficult to make a call over VoIP. In such a case, you can continue to open a network over PSTN. If PSTN backup is enabled, enable the busyout feature in the following method: Gateway(config-vservice-voip)# busyout monitor gatekeeper Gateway(config-vservice-voip)# busyout monitor callagent Gateway(config-vservice-voip)# busyout monitor voip-interface If the PSTN port exists at the VoIP gateway, the feature will not need to be enabled. If the gateway has only the FXS and FXO ports and you want to enable PSTN backup, execute the “PSTN-Backup-port” command at the FXS port. This feature enables the FXO port to operate just like the PSTN port. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.




[Figure 9] PSTN Backup When Inter-Working With Gatekeeper



Related Commands

Enabling PSTN-backup-port Setting

Step Command Description

1 #

# config

Switches to APOS command

setting mode.

2 (config)# voice-port 0/0 Starts voice port setting.

3 (config-voice-port-0/0)# pstn-backup-port <0-1/0-3> Sets up a PSTN backup port.

Disabling PSTN-backup-port Setting

Step Command Description 1 (config)#

(config)# no pstn-backup-port Disables the PSTN backup port.

Default: Disable



10. Restricting Call Duration

The “timeout tterm” command that restricts call duration has been executed for origination and reception as well as all ports of the VoIP gateway once the command is set. The “timeout tterm” command had been set as follows: (config)# voice service voip (config-vservice-voip)# timeout tterm <10 - 86400> The command has become to be executed in order to restrict only call duration at the target port. Since this command can be executed for any incoming and outgoing calls as well as any ports for VoIP services, you can use this command within the range you want. Note that the command is not executed and the timer does not operate until the caller lifts the handset. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 10] Restricting VoIP Gateway Call Duration



Related Commands

Enabling Timeout tteerm Feature


# config


setting mode.

2 (config)# voice-port 0/0 Moves to the voice port to be set.

3 (config-voice-port-0/0)# timeout tterm <10-86400> Enables the feature.

Disabling Timeout tterm Feature

Step Command Description 1 (config-voice-port-0/0)# no timeout tterm Disables the feature.

Default: Disable



11. Changing FAX Scenario In the past, a call has been disconnected automatically once a facsimile was sent; however, in v8.21 or later, a call is not disconnected and RTP continues to be used even after a facsimile is sent. You can have a conversation with the other party after a facsimile is sent. If necessary, the facsimile can be retransmitted. Any and all VoIP products of AddPac Technology support this feature. This feature is supported by the g711a/ulaw, g7231r63/53, and g729 codecs of the H323 protocol; however, is not supported by the g726r32/16 codecs and SIP.


[Figure 11] VoIP Gateway Fax Scenario

Default: Disable (A command that enables or disables the feature does not exist.)



12. busyout monitor callagent

If communications are not properly made with the gatekeeper due to a network failure or another error or a call cannot be delivered properly via the VoIP network, the gateway will provide the busyout monitoring service in a configuration that enables automatic transfer to PSTN. This command has been added to monitor binding with CallAgent. If binding with CallAgent is disconnected, execute the “busyout monitor callagent” command to enter busyout mode. The command inter-works with the “busyout-timer” command in mgcp setting mode. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


WAN(IP Network)

AddPacVoIP Gateway

Analog PhoneMGC

RSIPOK LAN Down

Off-HookNTFY

BusyoutStatus

voice service voipbusyout monitor callagent

busyout monitor timer timeout

[Figure 12] Busyout Monitor CallAgent



Related Commands

Enabling busyout monitor callagent Feature


# config


setting mode.

2 (config)# voice service voip Switches to VoIP service setting

mode.

3 (config-vservice-voip)# busyout monitor callagent Enables the feature.

Disabling busyout monitor callagent Feature

Step Command Description 1 (config-vservice-voip)# no busyout monitor callagent Disables the feature.

Default: Disable



13. rtp-nat-pat

If a gateway that has a private IP in the NAT/PAT environment attempts a call by using SIP, the call attempted from outside to the private gateway could not be made properly since the NAT/PAT server will not be able to know the RTP port used for the private gateway. In such a case, you can use the relevant commands to enable the NAT/PAT server to make a call properly by using the mapping table as follows: 1. The private gateway receives an INVITE message from outside. 2. The private gateway transfers the information on its RTP port to outside through the “rtp dummy packet” by using the INVITE message. 3. The NAT/PAT server reads the “rtp dummy packet” so that the mapping table can have the data on the RTP port used for the internal gateway. 4. If RTP is actually delivered from outside, the NAT/PAT server will use its mapping table to make a call properly. This feature has been implemented under SIP. The H323 protocol will support the feature. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 13] rtp-nat-pat Feature



Related Commands

Enabling rtp-nat-pat Feature


# config


setting mode.


mode.

3 (config-vservice-voip)# rtp-nat-pat Enables the feature.

Disabling rtp-nat-pat Feature

Step Command Description 1 (config-vservice-voip)# no rtp-nat-pat Disables the feature.

Default: Disable



14. SIP Call-transfer-mode [basic/attended]

AddPac Gateway supports a feature that enables the current VoIP call to be transferred to another user under SIP. AddPac Gateway supports Basic (Blind) Transfer mode and Attended Transfer mode as call transfer modes. You can check the difference between Basic Transfer mode and Attended Transfer mode from the figure below. The settings below are required to enable call transfer: (config)# dial-peer call-transfer h (config)# dial-peer call-hold h


WAN(IP Network)

AddPacVoIP Gateway

Analog Phone

AddPacVoIP Gateway

(Target)

(Transferor)

Analog Phone

SIP_Proxy

AddPacVoIP Gateway

(Transferee)

Analog Phone

200 OKINVITE

INVITE(HOLD)ACK

ACK200 OK

202 AcceptedREFER F1

200 OKNOTIFY (100 Trying) F2

200 OKBYE

Dial Tone

HookFlash

Dial Tone

Push Digit

Dial Tone

HookOnINVITE F3

200 OKACK

NOTIFY (200 OK) F4200 OK

Connect

[Figure 14] SIP Basic Call Transfer Mode



WAN(IP Network)

AddPacVoIP Gateway

Analog Phone

AddPacVoIP Gateway

(Target)

(Transferor)

Analog Phone

SIP_Proxy

AddPacVoIP Gateway

(Transferee)

Analog Phone

INVITE(h) / 200 / ACKINVITE / 200 / ACK

INVITE200 OK

REFER (Replaces)200 OK

Dial Tone

Hook-Flash

Dial Tone

Push Digit

Dial Tone

Hook-On

INVITE(Replaces)200 OK

ACK

Connect

ACKINVITE(h)

ACK200 OK

ACKBYE

200 OKNOTIFY (200 OK)

ACKBYE

[Figure 15] SIP Attended Call Transfer Mode

If you want to transfer a VoIP call while you are on the phone as shown in Figure 11, press the hook flash button on the phone. In such a case, the GW2 (Transferee) user will be on hold and hear a dial tone. The user presses the phone number of GW3 (Target) to be transferred. Then, the user will hear a tone, and operation methods are different in each call transfer mode. In Basic (Blind) Transfer mode, GW1 (Transferor) transfers a call to the GW3 user, and the GW1 user hangs up the phone. In Attended Transfer mode, the GW1 (Transferor) user transfers a call to GW3 (Target). Then, the GW3 user could make a call to the GW1 user if the GW3 user lifts the handset. This is a big difference between the two modes. The GW3 user may hang up the phone after the GW1 user transfers a call to the GW3 user.






Related Commands

Call-Transfer Mode Setting

Step Command Description 1 (config)# dial-peer call-hold h Enables the hook flash button to place a

call on hold.

2 (config)# dial-peer call-transfer h Enables the hook flash button to

transfer a call.

3 (config)# sip-ua Switches to sip-ua setting mode.

4 (config-sip-ua)# call-transfer <basic | attended> Sets call transfer mode.

Default: basic transfer



15. SIP Conference-server

A conference call allows more than two users to make a call. AddPac Gateway uses SIP to support 3-party conference calls. A separate conference server is required to enable this feature since AddPac Gateway itself does not support a conference call. It has been verified that a conference call is properly made by using Nortel Multimedia Communication Server (MCS) and Nortel Conference Server. The settings below are required to enable the conference call feature: (config)# dial-peer call-transfer h

(config)# dial-peer call-hold h The hook flash button on the phone is used to enable this feature. Press the hook flash button twice. This is the difference between call transfer. By default, the duration that AddPac Gateway recognizes the hook flash button is 500 ms. You should press the button for 500 ms (0.5 second) or less. If you think that 500 ms (0.5 second) is too short or the duration that PBX recognizes the hook flash button is 500 ms or more, you should change the hook flash detect timeout value. Refer to Step 4 of Related Commands. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.




[Figure 16] SIP Conference Call



Related Commands

Enabling Conference Call Feature


mode.

2 (config)# dial-peer call-hold h Enables the hook flash button to place a

call on hold.

3 (config)# dial-peer call-transfer h Enables the hook flash button to transfer a

call.

4 (config-vservice-voip)# timeout tdhf <500 - 3000> Sets the duration of recognizing the hook

flash button when a conference call is

made.


6 (config-sip-ua)# (config-sip-ua)# conference-server <server id>

Sets an ID in the conference server.

Disabling Polarity-Inverse Feature

Step Command Description 1 (config-sip-ua)# no conference-server Disables the conference server.

Default: Disable



16. SIP enable-ping

This feature is specific to Nortel. If AddPac Gateway operates in the PAT/NAT or firewall environment, a VoIP call will fail to be connected. A gateway in a private environment can exchange packets from an external public network through the PAT/NAT server; however, since a network device on an external public network cannot be aware of the IP and port No. of a gateway in a private environment, an incoming VoIP call cannot be made. To make incoming and outgoing VoIP calls in a private environment, you should set the public IP address of the PAT/NAT server at the gateway and enable the information on the VoIP port used for the gateway to be mapped with the PAT/NAT server statically. If you use Enable-Ping of Nortel to address this problem, you could make communications with SIP Proxy of Nortel and make an incoming/outgoing call to SIP Proxy properly. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.




[Figure 17] SIP Enable-Ping



Related Commands

Enabling Enable Ping Feature


# config


setting mode.


3 (config-sip-ua)# timeout tsipping <1 - 86400> Sets the cycle of forwarding the

ping packet.

3 (config-sip-ua)# enable-ping <entity-name> Performs SIP ping setting.

<entity-name>:

Sets an ID in the ping server.

Disabling Enable Ping Feature

Step Command Description 1 (config-sip-ua)# no enable-ping Disables the Enable Ping feature.

Default: Disable



17. SIP Media-channel [early|late]

The media channel refers to the RTP/RTPC route for making a VoIP communication. The media channel is categorized into the Tx channel and Rx channel for VoIP communications. This feature allows you to set a time when the Tx channel is enabled. If the gateway operates in the PAT/NAT environment and an inband (RTP) ringback tone (as well as color ring) is sent from the VoIP device of the other party, the inband ringback tone could not be delivered to the gateway in a private environment. The cause is the same as the one described in 12. SIP Enable-ping. To address this problem, open the Tx channel upon receiving 18X SDP after the gateway sends an INVITE message to the channel in a private environment. Then, a port table will be created at the PAT/NAT server so that you can hear an inband ringback tone. The media channel is categorized as follows: First, the default mode allows you to open the Tx channel only when receiving 183 Progress SDP after an INVITE message is sent. Even if 180 SDP is received, the Tx channel will not be open. Second, the early mode allows you to open the Tx channel when receiving 18X SDP after an INVITE message is sent. 18X SDP means that a 180 ring message includes SDP or a 183 progress message includes SDP. Third, the late mode allows you to open the Tx channel only when 200 OK is received after an INVITE message is sent. The Tx channel is not open when 18X SDP is received. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.




WAN(IP Network)

AddPacVoIP Gateway

Ethernet Switch Router(NAT Server)

Private IP Address: 10.1.1.9Default Gateway: 10.1.1.1Public IP: 192.168.1.9Medial-channel early/late/default

Private IP Address: 10.1.1.1Public IP Pool: 192.168.1.2 ~ 192.168.1.10Static Map for Private IP Address: 10.1.1.2 192.168.1.9

AddPacVoIP Gateway

Router

SIP Proxy

Public IP: 192.168.20.244

Public IP: 192.168.200.5

INVITE100 Trying

ACK

200 OK

Two Way RTP Media

INVITE18X (Ring/Progress)

18X (Ring/Progress)

200 OK

ACK

Analog Phone Analog Phone

Media-channel early

[Figure 18] SIP Media-Channel Early



Related Commands

Enabling Media Channel Feature


# config


setting mode.


3 (config-sip-ua)# media-channel <early | late> Changes media channel mode.

Setting Media Channel Feature to Default Mode

Step Command Description 1 (config-sip-ua)# no media-channel Switches to default mode.

Default: no media-channel



18. SIP Remote-party-id

If you want to apply the user name not E.164 defined in the destination pattern to the From field when an INVITE message is sent, enable this feature. AddPac Gateway supports a feature specific to Nortel. Typically, a caller ID refers to the user ID from the From field. If an IP phone inter-works with the trunk gateway at Nortel, a string will be used as the user ID and the caller ID will be converted to digits at SIP Proxy Server. In such a case, since SIP Proxy cannot convert the From field, it uses an option field, remote-party-id, to create, detect, or display the caller ID. This feature enables AddPac Gateway to inter-work with the VoIP device provided by Nortel. The Remote Party ID field is created in FXO, E&M, and E1/T1-type modules of AddPac Gateway not in the FXS-type module. Note that a remote party ID is not created in the FXS-type module even if this feature is enabled. All modules of AddPac Gateway recognize a caller ID by referring to the Remote Party ID optional field, if any, when receiving an INVITE message. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.




[Figure 19] SIP Remote-Party-ID



Related Commands

Enabling Remote-party-id Feature


# config


setting mode.


3 (config-sip-ua)# remote-party-id Creates a Remote-party-id field

in the INVITE message.

Disabling Remote-party-id Feature

Step Command Description 1 (config-sip-ua)# no remote-party-id Does not create a remote-party-

id field in the INVITE message.

Default: Disable



19. SIP route-by-auxiliary

AddPac Gateway uses the user ID of Request URI to perform dial-peer POTS peer routing when receiving an INVITE message. If the user ID of Request URI is [email protected], a call will be made to the port where the destination pattern of dial-peer is set to 1000. Enable this feature by referring to the user ID in the To field of the initial INVITE message if you want to perform routing. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.

Related Commands

Enabling Route-by-auxiliary Feature


# config


setting mode.


3 (config-sip-ua)# route-by-auxiliary Enables the Route-by-auxiliary

feature.

Disabling Route-by-auxiliary Feature

Step Command Description 1 (config-sip-ua)# route-by-auxiliary Disables the Route-by-auxiliary

feature.

Default: Disable



20. SIP Call-Diversion

SIP can enable the call diversion feature, which has been supported only by H.323. The SIP call diversion feature supported by AddPac Gateway enables a call to be transferred to another gateway when the recipient is on the phone, does not answer the call, or is in unconditional mode. If the recipient does not answer a call, a sound will ring for 15 seconds from the phone of the recipient and then call diversion will be performed. The unconditional mode enables any calls to be transferred to another gateway if the original gateway has a failure. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 20] SIP Call Diversion



Related Commands

Enabling Call Diversion


2 (config)# call-diversion 0 Switches to call diversion setting mode.

3 (config-call-diversion#0)# cfb-called e164 2003

(config-call-diversion#0)# cfnr-called e164 2003

(config-call-diversion#0)# cfu e164 2003

Sets a phone number to which a call will be

transferred when the recipient is on the

phone, does not answer a call, or is in

unconditional mode.

4 (config)# dial-peer voice 0 pots

(config-dialpeer-pots-0)# diversion 0

Enables call diversion in the POTS peer.

5 (config)# dial-peer voice 1002 voip

(config-dialpeer-voip-1002)# destination-pattern

2003

(config-dialpeer-voip-1002)# session protocol sip

(config-dialpeer-voip-1002)# session target

172.20.101.30

Creates a VoIP peer for e164 setup by call

diversion.

Note that the gateway where call diversion is enabled should have a dial-peer for the destination of a transferred call. This feature enables a call to be transferred by using the Contact value in the 302 Move Temporarily message. The Contact value is set in the form of “e164@IP”. Set e164 by referring to the call diversion setting, and set IP by referring to the session target IP of voip-peer. You should create a voip-peer for the gateway to which a call is transferred.



21. SIP user-config call-diversion

You can enable call diversion by using a specific code. The call diversion feature enabled by using a code value is performed the same as the one enabled in the console. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 21] cfb-activation Setting Using User Input



Related Commands

Enabling Input Code


mode.

2 (config)# dial-peer cfb-activation *99

(config)# dial-peer cfb-deactivation #99

(config)# dial-peer cfnr-activation *98

(config)# dial-peer cfnr-deactivation #98

(config)# dial-peer cfu-activation *97

(config)# dial-peer cfu-deactivation #97

Sets the user input code value to

enable call diversion.

Disabling Input Code

Step Command Description 1 (config)# no dial-peer cfb

(config)# no dial-peer cfnr

(config)# no dial-peer cfu

Disables the feature.

Default: Disable



22. SIP set-local-domain

If you want to set the URL of an SIP message to a specific domain not the IP address of AddPac Gateway, enable this feature. Typically, the IP address of AddPac Gateway is used as the URL of an SIP message as follows:

REGISTER sip:172.17.202.100 SIP/2.0

Via: SIP/2.0/UDP 172.17.201.51:5060;branch=z9hG4bK3064de00a41

From: <sip:[email protected]>;tag=3064de00a4

To: sip:[email protected]

Call-ID: [email protected]

CSeq: 1 REGISTER

Date: Sat, 11 Mar 2023 05:44:16 GMT

User-Agent: AddPac SIP Gateway

Contact: <sip:[email protected]>;expires=60

Expires: 60

Content-Length: 0

Max-Forwards: 70

You can use a domain as the URL of an SIP message as follows:

REGISTER sip:addpac.com SIP/2.0





CSeq: 2 REGISTER

Date: Sat, 11 Mar 2023 05:44:16 GMT



Expires: 60

Content-Length: 0

Max-Forwards: 70

If you want to use a domain as the URL of an SIP message, the domain must be registered with the SIP server so that the URL can be set to the domain. Note that the URL of an SIP message will be set to the IP address of AddPac Gateway if the domain is not registered with the SIP server or a peer to peer communication is made. Any and all VoIP products of AddPac Technology support this feature. You can enable



or disable the feature.

Related Commands

Enabling Set-local-domain Feature


# config


setting mode.


3 (config-sip-ua)# set-local-domain <string> Ex) set-local-domain sip.addpac.com

Sets a domain to be used for

AddPac Gateway.

Disabling Set-local-domain Feature

Step Command Description 1 (config-sip-ua)# no set-local-domain Uses the IP address of AddPac

Gateway as the URL of an SIP

message.

Default: Disable



23. SIP set-local-host

If you want to set the URL of an SIP message to a host name not the IP address of AddPac Gateway, enable this feature. Typically, the IP address of AddPac Gateway is used as the URL of an SIP message as follows:






CSeq: 1 REGISTER

Date: Sat, 11 Mar 2023 05:44:16 GMT



Expires: 60

Content-Length: 0

Max-Forwards: 70

You can use a host name as the URL of an SIP message as follows:



From: <sip:9000@AP200-hostname>;tag=3064de00a4

To: sip:9000@AP200-hostname


CSeq: 1 REGISTER

Date: Sat, 11 Mar 2023 05:44:16 GMT



Expires: 60

Content-Length: 0

Max-Forwards: 70




Related Commands

Enabling SIP set-local-host Feature


# config


setting mode.

2 (config)# sip-ua Switches to sip setting mode.

3 (config-sip-ua)# set-local-host Enables set-local-host.

Disabling SIP set-local-host Feature

Step Command Description 1 (config-sip-ua)# no set-local-host Initializes set-local-host.

Default: Disable



24. SIP SRV enable

To enable this feature, register DNS as follows: (config)# dnshost nameserver IP-Address Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 22] SIP DNS-SRV



Related Commands

Enabling DNS SRV Feature


2 (config)# sip-ua Enters SIP UA setting mode.

3 (config-sip-ua)# srv enable Enables the DNS SRV feature.

4 (config-sip-ua)# sip-server voip.addpac.com Sets the IP address of SIP Server to DNS.

5 (config-sip-ua)# sip user-name Addpac Registers the user name of SIP.

6 (config-sip-ua)# sip password 1234 Registers the password of SIP.

7 (config-sip-ua)# register e164 Registers E.164 in SIP Server.

8 (config-sip-ua)# exit Disables SIP UA setting mode.


Disabling DNS SRV Feature

Step Command Description 1 (config-sip-ua)# no srv Disables the DNS SRV feature.

Default: Disable



25. SIP rel1xx

The basic call procedure under SIP is as follows: A 1xx response to an INVITE message is delivered, and then ACK to the message is sent. If you want to establish a policy that delivers a 1xx response and then receives ACK to the response, enable this feature. Note that ACK to a 1xx response is supported but ACK to 100 Trying is not supported. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 23] SIP rel1xx



Related Commands

Enabling SIP rel1xx


2 (config)# sip-ua Enters Sip-ua setting mode.

3 (config-sip-ua)# rel1xx supported Enables support to 100rel.

4 (config-sip-ua)# exit Disables Sip-ua setting mode.


Initializing SIP rel1xx

Step Command Description 1 (config-sip-ua)# no rel1xx Initializes SIP rel1xx.

Default: Disable



26. SIP Response

This feature allows you to make a response to 180 Ringing / 183 Progress upon reception under SIP. Typically under SIP, 180 Ringing does not include SDP, and 183 Progress includes SDP. This feature has been added to support the recent common SIP Server/Trunk G.W. and multimedia session; thus, should be enabled depending on the environment. This feature supports the Early Offer/Answer model (with a selective provisional response message). Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 24] Support Early Offer/Answer model



Related Commands

Enabling SIP Responses


2 (config)# sip-ua Enters Sip-ua setting mode.

3 (config-sip-ua)# response alert without sdp Disables SDP if AddPac Gateway sends 180

Ringing (optional).

4 (config-sip-ua)# exit Disables sip-ua setting mode.


Disabling SIP Responses

Step Command Description 1 (config-sip-ua)# response default Initializes the SIP response setting.

Default: Disable



27. SIP dtmf-relay h245-signal

Dual-mode, which is one of the methods of transmitting dtmf-relay in an environment based on SIP, operates the digits entered by the user after a call is connected as long as the buttons are pressed based on RTP while delivering events in the form of an INFO message. If you want to deliver only an INFO message not RTP, execute the relevant commands. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 25] DTMF-Relay h245-signal



Related Commands

Enabling SIP dtmf-relay h245-signal


2 (config)# dial-peer voice 1000 voip Enters voip-peer setting mode.

3 (config-dialpeer-voip-1000)# dtmf-relay h245-signal Changes the dtmf-relay method to a h245-

signal.

4 (config-dialpeer-voip-1000)# session protocol sip Enables SIP.

5 (config-dialpeer-voip-1000)# exit Disables setting mode.

Disabling SIP dtmf-relay h245-signal

Step Command Description 1 (config-dialpeer-voip-1000)# dtmf-relay h245-

alphanumeric

Initializes dtmf-relay.

Default: Disable



28. Secondary MGC Registration

This feature has been added to switch over primary/secondary MGC even under MGCP just like the H.323 protocol. The secondary MGC can be registered by using priority. If the secondary MGC is not registered by using priority. The primary/secondary MGC is automatically sorted out depending on the sequence of registered call agents. Any and all VoIP products of AddPac Technology support this feature.


[Figure 26] VoIP Gateway Secondary MGC



Related Commands

Registering Secondary MGC


# config


setting mode.

2 (config)# mgcp Starts mgcp setting.

3 (config)# call-agent 10.1.1.1 2427 128 (config)# call-agent 20.1.1.1 2427 254

Registers the primary/secondary

MGC.

Deregistering Secondary MGC

Step Command Description 1 (config-mgcp)# no call-agent 20.1.1.1 Deregisters the secondary MGC.

Default: Disable



29. Individual MGCP Registration

In the past, MGC has been able to be registered or deregistered with or from several pots-peer at a time. This feature allows you to register or deregister MGC with or from one pots-peer. Any and all VoIP products of AddPac Technology support this feature.


[Figure 27] Registering MGCP new_user of VoIP Gateway



Related Commands

Individual MGCP Registration


# config


setting mode.

2 (config)# dial-peer voice 0 pots Starts pots-peer settings.

3 (config-dialpeer-pots-0)# no shutdown Attempts to register MGC.

Clearing Individual MGCP Registration

Step Command Description 1 (config-dialpeer-pots-0)# shutdown Attempts to deregister MGC.



30. MGCP busyout-timer

When you monitor binding with MGCP Call Agent, you could enable MGCP Call Agent to operate in busyout mode in a specific amount of time if MGC is not registered. To execute the relevant commands, enable call agent monitoring as follows: Gateway(config-vservice-voip)# busyout monitor callagent Verify if the timer starts once you lift the handset. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 28] VoIP Gateway MGCP busyout-timer



Related Commands

Enabling Busyout-timer


# config


setting mode.

2 (config)# mgcp Switches to mgcp setting mode.

3 (config-mgcp)# busyout-timer <1-30> Enables the busyout timer.

Disabling Busyout-timer

Step Command Description 1 (config-mgcp)# no busyout-timer Disables the busyout timer.

Default: Disable



31. MGCP dtmf-relay dual-mode

There are two digit delivery methods of dtmf-relay rtp-2833 and out-of-band to make a VoIP call using MGC. rtp-2833 enables a digit to play as long as the digit is pressed when the digit is entered after a call is made. Out-of-band transmits event logs on which digits have been pressed not how long a digit is pressed. An NTFY message including the relevant event log is forwarded to MGC. If you want to forward an NTFY message by playing a digit as long as the digit is pressed, enable dtmf-relay dual-mode. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 29] MGCP dtmf-relay dual-mode



Related Commands

Enabling dtmf-relay dual-mode


# config


setting mode.

2 (config)# mgcp Starts mgcp setting.

3 (config-mgcp)# (config-mgcp)# dtmf-relay dual-mode

Sets dtmf-relay dual-mode.

Disabling dtmf-relay

Step Command Description

1 (config-mgcp)# no dtmf-relay Disables the DTMF transmission

method.

Default: Disable



32. MGCP epid-type

Epid-type allows you to name an endpoint-id type. An epid-type should match the epid type required for authentication in MGC. Epid-type is categorized into two types as follows: First, the default type of an endpoint name is specified by using the information on slots/ports, and is displayed in the form of “aaln/slot/port@hostname”. Second, the specific type of an endpoint name is specified by using the counter information not slots/ports, and is displayed in the form of “aaln/counter@hostname”. Note that the counter starts from 1 not 0. A character specified such as “epid-type s# p#” can be added to the left of #. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 30] MGCP epid-type



Related Commands

Enabling FAX-Early-Detect


# config


setting mode.


3 (config-mgcp)# epid-type # Sets an endpoint type.

Disabling FAX-Early-Detect

Step Command Description 1 (config-mgcp)# epid-type # #

or (config-mgcp)# no epid-type

Initializes epid-type.

Default: Disable



33. FAX protocol ‘multi-session-t38’

A facsimile session opens after the re-invite message containing the information on “T38 CED Tone” is forwarded once “T38 CED Tone” arrives after the audio session opens. If you want to close the audio session and to maintain only the facsimile session when the audio session and facsimile session open simultaneously, you can enable the feature. If the audio and facsimile sessions are open simultaneously before ‘CED Tone’ is detected, it means that the audio session is activated and the facsimile session is on hold. In Version 8.23, this feature can be enabled only under MGCP. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 31] FAX protocol multi-session-t38



Related Commands

Enabling FAX protocol multi-session-t38


# config


setting mode.


mode.

3 (config-vservice-voip)# fax protocol multi-session-t38 Sets a facsimile protocol.

Disabling FAX protocol multi-session-t38

Step Command Description 1 (config-vservice-voip)# no fax protocol Initializes the facsimile protocol.

Default: Disable



34. MGCP force-local-rt

If an outgoing call is made via the VoIP gateway and the ‘rt’ data of the mdcx message received from MGC is received, a local ringback tone will be played. If the other party receives an inband ringback tone, the tone will be played. An inback ringback tone from the other party is mute, and a ringback tone plays back. Sometimes, a mute ringback tone plays back. In such a case, you can use the “force-local-rt” command. If an inband ringback tone is heard while a local ringback tone is playing back, ignore the inband ringback tone and continue to play back the local ringback tone before a call is connected. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


WAN(IP Network)

AddPacVoIP Gateway

Analog Phone Analog PhoneAddPacVoIP Gateway

MGC

notify

mgcp:force-local-rt

Inband Ringback Tone

Connect

dialingrqnt

mdcx: rtLocalRingback

ToneOff-Hook

[Figure 32] MGCP force-local-rt



Related Commands

Enabling Force-local-rt


# config


setting mode.


3 (config-mgcp)# force-local-rt Enables Force-local-rt.

Disabling Force-local-rt

Step Command Description 1 (config-mgcp)# no force-local-rt Disables Force-local-rt.

Default: Disable



35. E1/T1 compand-type [au-law|ua-law]

PBX has two types: a-law (Europe), which is the traditional PBX, and u-law (North America), which is a new PBX. When you use only a-law (Europe), a problem had not occurred while PBX is being used. As u-law (North America) was developed, PBX has conformed to u-law, and the E1 card contained in PBX has conformed to a-law. You can maintain the proper operation of PBX by adjusting the compand-type in the PCM mode/signal mode. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.

Related Commands

Enabling Compand-type


# config


setting mode.

2 (config)# voice-port 0/0 Moves to the voice port to be set.

3 (config-voice-port-0/0)# compand-type {au-law|ua-law} Sets Compand-type.

Disabling Compand-type

Step Command Description 1 (config-voice-port-0/0)# no compand-type Disables the Compand-type setting.

Default: Disable



36. E1/T1 isdn overlap-sending

Once the recipient sends setup ACK to the setup message received over VoIP, AddPac Gateway sends the setup ACK indicate data to the caller. In such a case, each character of a digit pressed by the caller is delivered to PBX in the form of information, and the PBX sends alert. An RTP communication path is established between the caller and recipient under H245. If a called number does not exist in the setup message sent from outside over VoIP, PBX itself will not be able to recognize DTMF and only the current flows while a call is connected. In such a case, execute the relevant commands to make a call properly. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.


[Figure 33] E1 PRI overlap-sending



Related Commands

Enabling isdn overlap-receiving


# config


setting mode.

2 (config)# controller e1 0/0 Moves to the E1 port to be set.

3 (config-controller-e1-0/0)# isdn overlap-sending Enables the overlap sending

feature.

Disabling isdn overlap-receiving

Step Command Description 1 (config-voice-port-0/0)# no isdn overlap-sending Disables the overlap sending feature.

Default: Disable



37. E1/T1 clock slave-main

This feature is used for a gateway model where 4E1 is installed. If the gateway enters in clock slave mode, a failure in facsimile reception may occur; thus, the slave-main setting is required for one controller port of 4E1. The slave-main setting can be performed only in one of four ports. Note that the slave setting is automatically performed in the other three ports when the slave-main setting is performed in one port. Any and all VoIP products of AddPac Technology support this feature. You can enable or disable the feature.

Related Commands

Enabling E1/T1 clock slave-main


# config


setting mode.

2 (config)# controller e1 0/0 Switches to E1 setting mode.

3 (config-controller-e1-0/0)# clock slave-main Enables the feature.

Initializing E1/T1 clock slave-main

Step Command Description 1 (config-controller-e1-0/0)# no clock Initializes the feature.

Default: Disable



38. ISDN PRI numbering-type

If the numbering type of the called/calling number among the setup messages received from the external (originating) gateway is PRI, it always will be a unknown type. To resolve this issue, the three types of commands below have been added. 1. isdn called(calling)-party-numbering-type { abbreviated | international | national | network | subscriber | unknown } 2. isdn called(calling)-party-numbering-type by-peer 3. isdn called(calling)-party-numbering-type from-network In No. 1, the numbering type defined in MG3000N PRI is applied to the setup message forwarded from MG3000N to PBX. In No. 2, the numbering type setting in POTS-peer/VoIP-peer is delivered to PBX as is. In No. 3, the numbering type is delivered to PBX by referring to the numbering type field of the setup message forwarded from the external (originating) gateway. Any and all VoIP products of AddPac Technology where the E1/T1 module can be installed support this feature. You can enable or disable the feature.


[Figure 34] ISDN numbering-type from-network



Related Commands

Enabling ISDN numbering-type


# config


setting mode.


3 (config-controller-e1-0/0)# isdn called-party-numbering-type from-network (config-controller-e1-0/0)# isdn calling-party-numbering-type from-network

Sets the numbering type of ISDN

on the network.

Disabling ISDN numbering-type

Step Command Description 1 (config-controller-e1-0/0)# isdn called-party-numbering-type

unknown (config-controller-e1-0/0)# isdn calling-party-numbering-type unknown

Initializes the numbering type of

ISDN.

Default: Disable



39. ISDN PRI numbering-plan

If the numbering type of the called number or calling number from the setup message received from the external (originating) gateway is PRI, it always will be a unknown type. Depending on the types of PBX, if an unknown numbering type of the called number or calling number is delivered, the setup message may not be able to be processed. To resolve this issue, the command below has been added: isdn called(calling)-party-numbering-plan {unknown|isdn-telephony |data|telex|national|private|from-network} Any and all VoIP products of AddPac Technology where the E1/T1 module can be installed support this feature. You can enable or disable the feature.


[Figure 35] ISDN numbering-type from-network



Related Commands

Enabling ISDN numbering-plan


# config


setting mode.


3 (config-controller-e1-0/0)# isdn called-party-numbering-plan data (config-controller-e1-0/0)# isdn calling-party-numbering-plan telex

Sets the numbering plan of ISDN.

Disabling ISDN numbering-plan

Step Command Description 1 (config-controller-e1-0/0)# isdn called-party-numbering-plan

unknown (config-controller-e1-0/0)# isdn calling-party-numbering-plan unknown

Initializes the numbering plan of

ISDN.

Default: Disable



40. ISDN immediate-disc In the past, AddPac Gateway has not been able to recognize a Disconnect message forwarded from PBX when VoIP calls of E1 ISDN flow via the PBX. In such a case, you can execute the relevant commands if you want to recognize a Disconnect message forwarded by PBX to AddPac Gateway and to deliver the message to the other party. Note that you should check if AddPac Gateway recognizes only Disconnect messages that have a progress indicator. Any and all VoIP products of AddPac Technology where the E1/T1 module can be installed support this feature. You can enable or disable the feature.


[Figure 36] ISDN immediate-disc Configuration



Related Commands

Enabling ISDN immediate


# config


setting mode.


3 (config-controller-e1-0/0)# isdn immediate-disc Sets ISDN immediate-disc.

Disabling ISDN immediate

Step Command Description 1 (config-controller-e1-0/0)# no isdn immediate-disc Initializes ISDN immediate-disc.

Default: Disable



41. R2 dtmf gain Change

If VoIP calls of E1 R2 flow via PBX and the PBX cannot recognize the dtmf gain value transmitted from AddPac Gateway, you can execute the relevant commands to resolve this issue. You can change the dtmf gain value to a higher one for proper dtmf transmittance. Any and all VoIP products of AddPac Technology where the E1/T1 module can be installed support this feature. You can enable or disable the feature.


[Figure 37] Changing R2 dtmf-gain Value



Related Commands

Changing R2 dtmf gain Value


# config


setting mode.

2 (config)# voice-port 0/0 Switches to voice-port setting

mode.

3 (config-voice-port-0/0)# high-dtmf-gain <-31 ~ 3> (config-voice-port-0/0)# low-dtmf-gain <-31 ~ 3>

Changes the dtmf-gain value.

Initializing R2 dtmf gain Value

Step Command Description 1 (config-voice-port-0/0)# no high-dtmf-gain

(config-voice-port-0/0)# no low-dtmf-gain Initializes the dtmf-gain value.



Removed Feature 42. Announcement The model below does not support the announcement feature: Model: AP1005



Troubleshooting43. Command Error, “clear-down-tone” Upon reboot, a cleardown tone used to be detected in relation to the detection of E1 and E&M cleardown tones by default. This error has been resolved. 44. Async Modem In the past, a VoIP call connected over Ethernet has been disconnected when the PPP session connection to the asynchronous modem fails (keep-active mode) in AP160. This error has been resolved. 45. Voip-interface If you set primary_voip_interface to async0 in AP160, set secondary_voip_interface to ethernet0.0, and use only LAN0, a VoIP call cannot be made. This error has been resolved. 46. Call Forwarding Errors When Ring Timer Expires When the ring timer expires after call forwarding no reply activation is enabled, a call is not forwarded. This error has been resolved. 47. PAT Table Display If the PAT table continues to display by executing the “show nat ether 1 0” command in the AP190 PAT environment, the device will reboot. This error has been resolved. 48. RTP Payload Timestamp Value Errors As the timestamp of RTP payload in DTMF-Relay RTP-2833 continues to increase, some terminals recognizes two tones. In such a case, fix the timestamp to troubleshoot this error. 49. STUN Disabled When IP Address is Changed If the IP_Address is changed while the STUN feature is enabled in an NAT environment, the feature will be disabled. This error has been resolved. 50. An INVITE Message Sent Before SIP Registration Through AP160 In the past, an INVITE message has been sent before SIP registration in an environment where AP160 Dialup Modem is used. This error has been resolved.


Documents

Release note VOIPv8.12 - addpac.su · APOS Release Note Voice over IP Added Features 1. STUN STUN (Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators