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SIP Timer Values (sec)

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SIP page Remove Last Reg Use Compact Header Escape Display Name RFC 2543 Call Hold Mark All AVT Packets SIP TCP Port Min SIP TCP Port Max Lets you remove the last registration before registering a new one if the value is different. Select yes or no from the drop-down menu. The default is no. Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the ATA device uses compact SIP headers in outbound SIP messages. If set to no, the ATA device uses normal SIP headers. If inbound SIP requests contain compact headers, ATA device reuses the same compact headers when generating the response regardless the settings of the Use Compact Header parameter. If inbound SIP requests contain normal headers, ATA device substitutes those headers with compact headers (if defined by RFC 261) if Use Compact Header parameter is set to yes. The default is no. Lets you keep the Display Name private. Select yes if you want the ATA device to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside the pair of double quotes. Otherwise, select no. The default is no. Configures the type of call hold: a:sendonly or 0.0.0.0. The default is no; do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax. If set to yes, all AVT tone packets (encoded for redundancy) have the marker bit set. If set to no, only the first packet has the marker bit set for each DTMF event. The default is yes. Specifies the lowest TCP port number that can be used for SIP sessions. This field is not found in the PAP2T. Specifies the highest TCP port number that can be used for SIP sessions. This field is not found in the PAP2T. Voice tab > SIP page > SIP Timer Values (sec) section Field SIP T1 SIP T2 SIP T4 SIP Timer B SIP Timer F Description RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds. The default is.5. RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds. The default is 4. RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds. The default is 5. INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. Non-INVITE time-out value, which can range from 0 to 64 seconds. The default is 32. Linksys ATA Administration Guide 103

Section	Page
Document Audience	8
Supported Firmware	9
Document Conventions	9
Document Purpose and Contents	10
Related Documentation	11
Online Resources	12
Copyright and Trademarks	12
Technical Support	12
Finding Information in PDF Files	13
Finding Text in a PDF	13
Finding Text in Multiple PDF Files	13
Introducing Linksys Analog Telephone Adapters	15
Comparison of ATA Devices	16
Linksys ATA Connectivity Requirements	18
Linksys PAP2T Connectivity	18
Linksys SPA2102 Connectivity	19
Linksys SPA3102 Connectivity	20
Linksys SPA8000 Connectivity	21
ATA Software Features	22
Voice Supported Codecs	22
SIP Proxy Redundancy	23
Other Linksys ATA Software Features	23
Basic Administration and Configuration of Your Linksys ATA Device	28
Basic Services and Equipment Required	28
Downloading Firmware	29
Basic Installation and Configuration	30
Upgrading the Firmware for the Linksys ATA Device	30
Setting up Your Linksys ATA Device	31
Using the Administration Web Server	31
Connecting to the Administration Web Server	32
Setting Up the WAN Configuration for Your Linksys ATA Device	32
Registering to the Service Provider	33
Advanced Configurations	34
Upgrading, Rebooting, and Resyncing Your Linksys ATA Device	34
Upgrade URL	34
Resync URL	35
Reboot URL	35
Provisioning Your Linksys ATA Device	35
Provisioning Capabilities	36
Configuration Profile	36
Configuring Your System for ITSP Interoperability	38
Network Address Translation (NAT) and Voice over IP (VoIP)	38
NAT Mapping with Session Border Controller	38
NAT Mapping with SIP-ALG Router	38
Configuring NAT Mapping with a Static IP Address	39
Configuring NAT Mapping with STUN	40
Determining Whether the Router Uses Symmetric or Asymmetric NAT	41
Firewalls and SIP	42
Configuring SIP Timer Values	43
Configuring Voice Services	44
Supported Codecs	44
Using a FAX Machine (SPA2102, SPA3102 or SPA8000)	45
Fax Troubleshooting	46
Managing Caller ID Service	47
Silence Suppression and Comfort Noise Generation	48
Configuring Dial Plans	49
About Dial Plans	49
Editing Dial Plans	55
Secure Call Implementation	57
Enabling Secure Calls	57
Secure Call Details	58
Using a Mini-Certificate	58
Generating a Mini Certificate	59
SIP Trunking and Hunt Groups on the SPA8000	61
About SIP Trunking	62
Setting the Trunk Group Call Capacity	64
Inbound Call Routing for a Trunk Group	64
Contact List for a Trunk Group	65
Outgoing Call Routing for a Trunk Group	66
Configuring a Trunk Group	67
Trunk Group Management	68
Setting the Hunt Policy	69
Additional Notes About Trunk Groups	69
Configuring Music on Hold	70
Using the Internal Music Source for Music On Hold	70
Using the Internal Music Source	70
Changing the Music File for the Internal Music Source	71
Configuring a Streaming Audio Server	71
About the Streaming Audio Server	71
Configuring the Streaming Audio Server	72
Using the IVR with an SAS Line	73
Configuring the PSTN (FXO) Gateway	74
Connecting to PSTN and VoIP Services	74
How VoIP-To-PSTN Calls Work	75
One-Stage Dialing	75
Two-Stage Dialing	76
How PSTN-To-VoIP Calls Work	77
Terminating Gateway Calls	77
VoIP Outbound Call Routing	78
Configuring VoIP Failover to PSTN	79
Sharing One VoIP Account Between the FXS and PSTN Lines	80
Other Options	81
PSTN Call to Ring Line 1	81
Symmetric RTP	81
Call Progress Tones	81
Call Scenarios	82
PSTN to VoIP Call with and Without Ring-Thru	82
VoIP to PSTN Call With and Without Authentication	83
Call Forwarding to PSTN Gateway	85
Linksys ATA Routing Field Reference	86
Status page	86
Product Information section	86
System Status section	87
WAN Setup page	87
Internet Connection Settings section	88
Static IP Settings section	88
PPPoE Settings section	88
Optional Settings section	89
MAC Clone Settings section	89
Remote Management section	90
QOS Settings section	90
VLAN Settings section	90
LAN Setup page	90
Networking Service section	91
LAN Networking Settings section	91
Static DHCP Lease Settings section	91
Application page	92
Port Forwarding Settings section	92
DMZ Settings section	92
Miscellaneous Settings section	93
System Reserved Ports Range section	93
Linksys ATA Voice Field Reference	94
Info page	94
Product Information section	95
System Status section	95
Line Status section	95
System Information section (PAP2T)	97
PSTN Line Status section (AG310 and SPA3102)	97
Trunk Status section (SPA8000)	99
System page	99
System Configuration section	100
Internet Connection Type section (PAP2T)	100
Optional Network Configuration section (PAP2T)	100
Miscellaneous Settings section (not used with PAP2T)	101
SIP page	102
SIP Parameters section	102
SIP Timer Values (sec) section	103
Response Status Code Handling section	105
RTP Parameters section	105
SDP Payload Types section	106
NAT Support Parameters section	107
Trunking Parameters section (SPA8000)	109
Regional page	110
Call Progress Tones section	110
Distinctive Ring Patterns section	112
Distinctive Call Waiting Tone Patterns section	113
Distinctive Ring/CWT Pattern Names section	113
Ring and Call Waiting Tone Spec section	114
Control Timer Values (sec) section	114
Vertical Service Activation Codes section	116
Vertical Service Announcement Codes section (SPA2102, SPA8000)	119
Outbound Call Codec Selection Codes section	120
Miscellaneous section	121
Line page	123
Line Enable section	124
Streaming Audio Server (SAS) section	124
NAT Settings section	125
Network Settings section	125
SIP Settings section	126
Call Feature Settings section	128
Proxy and Registration section	128
Subscriber Information section	130
Supplementary Service Subscription section	130
Audio Configuration section	132
Gateway Accounts section (SPA3102/AG310)	132
VoIP Fallback to PSTN section (SPA3102/AG310)	133
Dial Plan section	133
FXS Port Polarity Configuration section	135
Trunk Group page (SPA8000)	135
Line Enable section	135
Network Settings section	136
SIP Settings section	136
Subscriber Information section	138
Dial Plan section	139
NAT Settings section	139
Proxy and Registration section	140
PSTN Line page (AG310 and SPA3102)	141
Line Enable section	142
NAT Settings section	142
Network Settings section	142
SIP Settings section	143
Proxy and Registration section	145
Subscriber Information section	146
Audio Configuration section	146
Dial Plans section	149
VoIP-To-PSTN Gateway Setup section	149
VoIP Users and Passwords (HTTP Authentication) section	150
Ring Settings section	151
FXO (PSTN) Timer Values (sec) section	151
PSTN Disconnect Detection section	153
International Control (Settings) section	155
User page	156
Call Forward Settings section	157
Selective Call Forward Settings section	157
Speed Dial Settings section	158
Supplementary Service Settings section	158
Distinctive Ring Settings section	159
Ring Settings section	159
PSTN User page (AG310 and SPA3102)	160
PSTN-To-VoIP Selective Call Forward Settings section	160
PSTN-To-VoIP Speed Dial Settings section	161
PSTN Ring Thru Line 1 Distinctive Ring Settings section	161
PSTN Ring Thru Line 1 Ring Settings section	161
Provisioning Reference (WRP400)	162
Troubleshooting	171
Environmental Specifications	174
PAP2T	174
SPA2102	174
SPA3102	175
SPA8000	175
RTP300	176
WRP400	176
WRTP54G	177
AG310	177
Warranty Information	178
Limited Warranty	178
Exclusions and Limitations	178
Obtaining Warranty Service	179
Technical Support	179
Regulatory Information	180
Federal Communications Commission Interference Statement	180
Industry Canada Statement	180
Règlement d’Industry Canada	180
EC Declaration of Conformity (Europe)	181
User Information for Consumer Products Covered by EU Directive 2002/96/EC on Waste Electric and Electronic Equipment (WEEE)	181
Safety Information	188
Meaning of the Warning Symbol	188
General Safety Information	188
Power Safety Information	189
Software License Agreement	190
Software in Linksys Products:	190
Software Licenses:	190
Schedule 1 Linksys Software License Agreement	190
Schedule 2	192
Schedule 3	197

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Linksys ATA Administration Guide

103

SIP page

Voice tab > SIP page >

SIP Timer Values (sec) section

Remove Last Reg

Lets you remove the last registration before registering a new one if the

value is different. Select yes or

no from the drop-down menu.

The default is

Use Compact Header

Lets you use compact SIP headers in outbound SIP messages. Select yes

no from the drop-down menu. If set to yes, the ATA device uses compact SIP

headers in outbound SIP messages. If set to no, the ATA device uses normal

SIP headers. If inbound SIP requests contain compact headers, ATA device

reuses the same compact headers when generating the response regardless

the settings of the

Use Compact Header

parameter. If inbound SIP requests

contain normal headers, ATA device substitutes those headers with compact

headers (if defined by RFC 261) if

Use Compact Header

parameter is set to yes.

The default is

Escape Display Name

Lets you keep the Display Name private. Select yes if you want the ATA

device to enclose the string (configured in the Display Name) in a pair of

double quotes for outbound SIP messages. Any occurrences of or \ in the

string is escaped with \ and \\ inside the pair of double quotes. Otherwise,

select no.

The default is

RFC 2543 Call Hold

Configures the type of call hold: a:sendonly or 0.0.0.0.

The default is

; do not use the 0.0.0.0 syntax in a HOLD SDP; use the

a:sendonly syntax.

Mark All AVT Packets

If set to yes, all AVT tone packets (encoded for redundancy) have the marker

bit set. If set to no, only the first packet has the marker bit set for each DTMF

event.

The default is

yes

SIP TCP Port Min

Specifies the lowest TCP port number that can be used for SIP sessions. This

field is not found in the PAP2T.

SIP TCP Port Max

Specifies the highest TCP port number that can be used for SIP sessions. This

field is not found in the PAP2T.

Field

Description

SIP T1

RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds.

The default is

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests

and INVITE responses), which can range from 0 to 64 seconds.

The default is

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network),

which can range from 0 to 64 seconds.

The default is

SIP Timer B

INVITE time-out value, which can range from 0 to 64 seconds.

The default is

SIP Timer F

Non-INVITE time-out value, which can range from 0 to 64 seconds.

The default is