APOS Release Note

Transcription

APOS Release Note

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
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Voice over IP
[ 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
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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
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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
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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
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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.
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Network Configuration
Private IP Address: 10.1.1.1
Public IP Pool: 192.168.1.2 ~ 192.168.1.10
WAN
(IP Network)
AddPac
Router (NAT-Server)
Analog Phone
(1000)
AddPac
Stun
Server
VoIP Gateway
IP Address192.168.50.100
IP Address 10.1.1.2
Public IP
192.168.1.2
Gatekeeper
IP Address 192.168.2.200
Binding Request
Binding Response
RRQ
RCF
Local IP
Changed
1000 Alias
192.168.1.2
Binding Request
Public IP 192.168.1.3
Retry Timer
Binding Response
RRQ
RCF
1000 Alias
192.168.1.3
Binding Request
Binding Response
[Figure 1]
VoIP Gateway STUN Feature
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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.
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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
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Related Commands
Enabling udp-checksum
Step
Command
Description
1
# config
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
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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.
voice service voip:
qos-threshold delay 500
qos-threshold jitter 300
qos-threshold packet-loss 100
SNMP Management
WAN
(IP Network)
Analog Phone
AddPac
VoIP Gateway-A
AddPac
VoIP Gateway-B
Analog Phone
GET SNMPv2-MIB
RESPONSE SNMPv2-MIB
[Figure 3]
Sending TOS Information When a Request for SNMP Get is Made
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Related Commands
Enabling QOS-Threshold
Step
Command
Description
1
# config
Switches to APOS command setting
mode.
2
(config)# voice service voip
3
(config-vservice-voip)# qos-threshold delay 1000
Sets qos-threshold.
(config-vservice-voip)# qos-threshold jitter 500
(config-vservice-voip)# qos-threshold packet-loss 5
Initializing QOS-Threshold
Step
Command
Description
1
(config-vservice-voip)# no qos-threshold delay
Initializes qos-threshold.
(config-vservice-voip)# no qos-threshold jitter
(config-vservice-voip)# no qos-threshold packet-loss
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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’.
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[Figure 4]
VoIP Gateway PPTP
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Related Commands
Enabling PPTP Route Data Feature
Step
Command
Description
1
# config
2
(config)# interface ether 0.0
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-
Sets an IP address. (For information on how
MASK
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-
Allows you to transfer data to a public
if ether0.0
network and to send VoIP traffic to a private
network.
18
(config)# exit
Disables setting mode.
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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
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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”.
Any and all VoIP products of AddPac Technology support this feature. You can enable
or disable the feature.
[Figure 5]
Sending VoIP Gateway Connect Message
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Related Commands
Enabling delayed-connect Feature
Step
Command
Description
1
#
# config
mode.
2
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
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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.
[Figure 6]
Changing CID Gain of VoIP Gateway
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Related Commands
Changing caller-id gain Value
Step
Command
Description
1
#
# 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.
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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
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Related Commands
Enabling voice-confirmed-connect timeout Feature
Step
Command
Description
1
#
# config
mode.
2
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
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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.
voice-service-voip
Timeout tttl <10-86400> fixed
WAN
(IP Network)
Analog Phone
GateKeeper
AddPac
VoIP Gateway
RRQ
RCF
ttl timeout
Gatekeeper
TTL timeout
[Figure 8]
RRQ
RCF
VoIP Gateway TTL Timeout
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Related Commands
Enabling ttl timer fixed Feature
Step
Command
Description
1
#
# config
mode.
2
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
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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.
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[Figure 9]
PSTN Backup When Inter-Working With Gatekeeper
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Related Commands
Enabling PSTN-backup-port Setting
Step
Command
Description
#
Switches to APOS command
# config
setting mode.
2
Starts voice port setting.
3
(config-voice-port-0/0)# pstn-backup-port <0-1/0-3>
Sets up a PSTN backup port.
1
Disabling PSTN-backup-port Setting
Step
Command
Description
1
(config)#
(config)# no pstn-backup-port
Disables the PSTN backup port.
Default: Disable
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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-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.
[Figure 10]
Restricting VoIP Gateway Call Duration
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Related Commands
Enabling Timeout tteerm Feature
Step
Command
Description
1
#
# config
setting mode.
2
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
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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.)
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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.
voice service voip
busyout monitor callagent
WAN
(IP Network)
Analog Phone
AddPac
VoIP Gateway
MGC
RSIP
OK
Off-Hook
Busyout
Status
[Figure 12]
LAN Down
NTFY
busyout monitor
timer timeout
Busyout Monitor CallAgent
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Related Commands
Enabling busyout monitor callagent Feature
Step
Command
Description
1
#
# config
setting mode.
Switches to VoIP service setting
2
mode.
3
(config-vservice-voip)# busyout monitor callagent
Disabling busyout monitor callagent Feature
Step
Command
Description
1
(config-vservice-voip)# no busyout monitor callagent
Default: Disable
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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.
[Figure 13]
rtp-nat-pat Feature
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Related Commands
Enabling rtp-nat-pat Feature
Step
Command
Description
1
#
# config
setting mode.
2
mode.
3
(config-vservice-voip)# rtp-nat-pat
Disabling rtp-nat-pat Feature
Step
Command
Description
1
(config-vservice-voip)# no rtp-nat-pat
Default: Disable
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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
SIP_Proxy
WAN
(IP Network)
AddPac Analog Phone
VoIP Gateway
(Target)
AddPac Analog Phone
VoIP Gateway
(Transferee)
Analog Phone AddPac
VoIP Gateway
(Transferor)
HookFlash
Dial Tone
Push Digit
Dial Tone
HookOn
Dial Tone
INVITE
200 OK
ACK
INVITE(HOLD)
200 OK
ACK
REFER F1
202 Accepted
NOTIFY (100 Trying) F2
200 OK
BYE
200 OK
INVITE F3
200 OK
ACK
NOTIFY (200 OK) F4
200 OK
Connect
[Figure 14]
SIP Basic Call Transfer Mode
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SIP_Proxy
WAN
(IP Network)
AddPac Analog Phone
VoIP Gateway
(Target)
AddPac Analog Phone
VoIP Gateway
(Transferee)
Analog Phone AddPac
VoIP Gateway
(Transferor)
HookFlash
Dial Tone
Push Digit
INVITE / 200 / ACK
INVITE(h) / 200 / ACK
Dial Tone
INVITE
200 OK
ACK
INVITE(h)
200 OK
ACK
Hook-On
Dial Tone
REFER (Replaces)
200 OK
INVITE(Replaces)
200 OK
ACK
BYE
ACK
NOTIFY (200 OK)
200 OK
BYE
ACK
Connect
[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.
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Related Commands
Call-Transfer Mode Setting
Step
Command
Description
1
Enables the hook flash button to place a
call on hold.
2
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
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Voice over IP
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:
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.
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Voice over IP
[Figure 16]
SIP Conference Call
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Related Commands
Enabling Conference Call Feature
Step
Command
Description
1
# config
mode.
2
Enables the hook flash button to place a
call on hold.
3
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.
5
(config)# sip-ua
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
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Voice over IP
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.
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[Figure 17]
SIP Enable-Ping
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Related Commands
Enabling Enable Ping Feature
Step
Command
Description
1
#
# config
setting mode.
2
(config)# sip-ua
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
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Voice over IP
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.
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Public IP Pool: 192.168.1.2 ~ 192.168.1.10
Static Map for Private IP Address: 10.1.1.2
192.168.1.9
Public IP: 192.168.200.5
Ethernet Switch
Router
(NAT Server)
Analog Phone AddPac
WAN
(IP Network)
Router
AddPac Analog Phone
VoIP Gateway
VoIP Gateway
Default Gateway: 10.1.1.1
Public IP: 192.168.1.9
Medial-channel early/late/default
SIP Proxy
Public IP: 192.168.20.244
INVITE
100 Trying
Media-channel early
INVITE
18X (Ring/Progress)
18X (Ring/Progress)
200 OK
200 OK
ACK
ACK
Two Way RTP Media
[Figure 18]
SIP Media-Channel Early
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Related Commands
Enabling Media Channel Feature
Step
Command
Description
1
#
# config
setting mode.
2
(config)# sip-ua
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
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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.
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[Figure 19]
SIP Remote-Party-ID
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Related Commands
Enabling Remote-party-id Feature
Step
Command
Description
1
#
# config
setting mode.
2
(config)# sip-ua
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-partyid field in the INVITE message.
Default: Disable
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Voice over IP
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.
Related Commands
Enabling Route-by-auxiliary Feature
Step
Command
Description
1
#
# config
setting mode.
2
(config)# sip-ua
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
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Voice over IP
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.
[Figure 20]
SIP Call Diversion
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Related Commands
Enabling Call Diversion
Step
Command
Description
1
# config
2
(config)# call-diversion 0
Switches to call diversion setting mode.
3
(config-call-diversion#0)# cfb-called e164 2003
Sets a phone number to which a call will be
(config-call-diversion#0)# cfnr-called e164 2003
transferred when the recipient is on the
(config-call-diversion#0)# cfu e164 2003
phone, does not answer a call, or is in
unconditional mode.
4
Enables call diversion in the POTS peer.
(config)# dial-peer voice 0 pots
(config-dialpeer-pots-0)# diversion 0
5
(config)# dial-peer voice 1002 voip
Creates a VoIP peer for e164 setup by call
(config-dialpeer-voip-1002)#
diversion.
destination-pattern
2003
(config-dialpeer-voip-1002)# session protocol sip
(config-dialpeer-voip-1002)# session target
172.20.101.30
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.
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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.
[Figure 21]
cfb-activation Setting Using User Input
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Related Commands
Enabling Input Code
Step
Command
Description
1
# config
mode.
2
(config)# dial-peer cfb-activation *99
Sets the user input code value to
(config)# dial-peer cfb-deactivation #99
enable call diversion.
(config)# dial-peer cfnr-activation *98
(config)# dial-peer cfnr-deactivation #98
(config)# dial-peer cfu-activation *97
(config)# dial-peer cfu-deactivation #97
Disabling Input Code
Step
Command
Description
1
(config)# no dial-peer cfb
(config)# no dial-peer cfnr
(config)# no dial-peer cfu
Default: Disable
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Voice over IP
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.
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Related Commands
Enabling Set-local-domain Feature
Step
Command
Description
1
#
# config
setting mode.
2
(config)# sip-ua
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
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Voice over IP
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
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Voice over IP
Related Commands
Enabling SIP set-local-host Feature
Step
Command
Description
1
#
# 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
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Voice over IP
24. SIP SRV enable
To enable this feature, register DNS as follows:
(config)# dnshost nameserver IP-Address
[Figure 22]
SIP DNS-SRV
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Related Commands
Enabling DNS SRV Feature
Step
Command
Description
1
# config
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.
9
(config)# exit
Disabling DNS SRV Feature
Step
Command
Description
1
(config-sip-ua)# no srv
Disables the DNS SRV feature.
Default: Disable
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Voice over IP
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.
[Figure 23]
SIP rel1xx
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Related Commands
Enabling SIP rel1xx
Step
Command
Description
1
# config
2
(config)# sip-ua
Enters Sip-ua setting mode.
3
(config-sip-ua)# rel1xx supported
Enables support to 100rel.
4
Disables Sip-ua setting mode.
5
(config)# exit
Initializing SIP rel1xx
Step
Command
Description
1
(config-sip-ua)# no rel1xx
Initializes SIP rel1xx.
Default: Disable
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Voice over IP
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).
[Figure 24]
Support Early Offer/Answer model
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Related Commands
Enabling SIP Responses
Step
Command
Description
1
# config
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
Disables sip-ua setting mode.
5
(config)# exit
Disabling SIP Responses
Step
Command
Description
1
(config-sip-ua)# response default
Initializes the SIP response setting.
Default: Disable
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Voice over IP
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.
[Figure 25]
DTMF-Relay h245-signal
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Related Commands
Enabling SIP dtmf-relay h245-signal
Step
Command
Description
1
# config
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 h245signal.
4
(config-dialpeer-voip-1000)# session protocol sip
Enables SIP.
5
(config-dialpeer-voip-1000)# exit
Disabling SIP dtmf-relay h245-signal
Step
Command
1
(config-dialpeer-voip-1000)#
Description
dtmf-relay
h245-
Initializes dtmf-relay.
alphanumeric
Default: Disable
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Voice over IP
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
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Related Commands
Registering Secondary MGC
Step
Command
Description
1
#
# config
setting mode.
2
(config)# mgcp
Starts mgcp setting.
(config)# call-agent 10.1.1.1 2427 128
(config)# call-agent 20.1.1.1 2427 254
Registers the primary/secondary
3
MGC.
Deregistering Secondary MGC
Step
Command
Description
1
(config-mgcp)# no call-agent 20.1.1.1
Deregisters the secondary MGC.
Default: Disable
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Voice over IP
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
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Related Commands
Individual MGCP Registration
Step
Command
Description
1
#
setting mode.
# config
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.
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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.
[Figure 28]
VoIP Gateway MGCP busyout-timer
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Related Commands
Enabling Busyout-timer
Step
Command
Description
1
#
# 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
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Voice over IP
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.
[Figure 29]
MGCP dtmf-relay dual-mode
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Related Commands
Enabling dtmf-relay dual-mode
Step
Command
Description
1
#
# 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
1
(config-mgcp)# no dtmf-relay
Description
Disables the DTMF transmission
method.
Default: Disable
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Voice over IP
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 #.
[Figure 30]
MGCP epid-type
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Related Commands
Enabling FAX-Early-Detect
Step
Command
Description
1
#
# config
setting mode.
2
(config)# mgcp
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
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Voice over IP
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.
[Figure 31]
FAX protocol multi-session-t38
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Related Commands
Enabling FAX protocol multi-session-t38
Step
Command
Description
1
#
# config
setting mode.
2
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
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Voice over IP
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.
MGC
mgcp:
force-local-rt
WAN
(IP Network)
Analog Phone
AddPac Analog Phone
VoIP Gateway
AddPac
VoIP Gateway
dialing
Local
Ringback
Tone
notify
rqnt
mdcx: rt
Inband Ringback Tone
Off-Hook
Connect
[Figure 32]
MGCP force-local-rt
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APOS Release Note
Voice over IP
Related Commands
Enabling Force-local-rt
Step
Command
Description
1
#
# config
setting mode.
2
(config)# mgcp
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
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APOS Release Note
Voice over IP
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.
Related Commands
Enabling Compand-type
Step
Command
Description
1
#
# config
setting mode.
2
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
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APOS Release Note
Voice over IP
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.
[Figure 33]
E1 PRI overlap-sending
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APOS Release Note
Voice over IP
Related Commands
Enabling isdn overlap-receiving
Step
Command
Description
1
#
# 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
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APOS Release Note
Voice over IP
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.
Related Commands
Enabling E1/T1 clock slave-main
Step
Command
Description
1
#
# config
setting mode.
2
Switches to E1 setting mode.
3
(config-controller-e1-0/0)# clock slave-main
Initializing E1/T1 clock slave-main
Step
Command
Description
1
(config-controller-e1-0/0)# no clock
Initializes the feature.
Default: Disable
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APOS Release Note
Voice over IP
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
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APOS Release Note
Voice over IP
Related Commands
Enabling ISDN numbering-type
Step
Command
Description
1
#
# config
setting mode.
2
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
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APOS Release Note
Voice over IP
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}
[Figure 35]
ISDN numbering-type from-network
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APOS Release Note
Voice over IP
Related Commands
Enabling ISDN numbering-plan
Step
Command
Description
1
#
# config
setting mode.
2
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
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APOS Release Note
Voice over IP
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.
[Figure 36]
ISDN immediate-disc Configuration
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APOS Release Note
Voice over IP
Related Commands
Enabling ISDN immediate
Step
Command
Description
1
#
# config
setting mode.
2
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
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APOS Release Note
Voice over IP
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.
[Figure 37]
Changing R2 dtmf-gain Value
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APOS Release Note
Voice over IP
Related Commands
Changing R2 dtmf gain Value
Step
Command
Description
1
#
# config
setting mode.
Switches to voice-port setting
2
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.
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APOS Release Note
Voice over IP
Removed Feature
42. Announcement
The model below does not support the announcement feature:
Model: AP1005
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APOS Release Note
Voice over IP
Troubleshooting
43. 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.
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APOS Release Note

Transcription

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