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Cisco Linksys SPA3102, Administration Manual

The Cisco Linksys SPA3102, a versatile VoIP telephone adapter, allows seamless integration of your traditional phone line with Internet telephony services. Unlock its full potential by accessing its detailed user manual, available for free download at our website. FAQs, troubleshooting tips, and step-by-step instructions await you at manualshive.com.

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Introducing Cisco Small Business Analog Telephone Adapters

ATA Software Features

Cisco Small Business ATA Administration Guide

30

1

 

Secure Calls

A user (if enabled by service provider or administrator) 
has the option to make an outbound call secure in the 
sense that the audio packets in both directions are 
encrypted. See 

“Secure Call Implementation” section 

on page 71

.

Adjustable Audio 
Frames Per Packet

This feature allows the user to set the number of audio 
frames contained in one RTP packet. Packets can be 
adjusted to contain from 1–10 audio frames. Increasing the 
number of packets decreases the bandwidth utilized, but 
it also increases delay and may affect voice quality. See 
the RTP Packet Size parameter found in the SIP tab in the 

“ATA Voice Field Reference,” on page 115

.

DTMF

The ATA device may relay DTMF digits as out-of-band 
events to preserve the fidelity of the digits. This can 
enhance the reliability of DTMF transmission required by 
many IVR applications such as dial-up banking and airline 
information. DTMF is configured in the 

DTMF Tx Mode

 

parameter found in the Line tabs. See the 

“ATA Voice Field 

Reference,” on page 115

.

Call Progress Tone 
Generation

The ATA device has configurable call progress tones. Call 
progress tones are generated locally on the ATA device so 
an end user is advised of status (such as ringback). 
Parameters for each type of tone (for instance a dial tone 
played back to an end user) may include frequency and 
amplitude of each component, and cadence information. 
See the Regional tab in the 

“ATA Voice Field Reference,” 

on page 115

.

Call Progress Tone 
Pass Through

This feature allows the user to hear the call progress tones 
(such as ringing) that are generated from the far-end 
network. See the Regional tab in the 

“ATA Voice Field 

Reference,” on page 115

.

Echo Cancellation

Impedance mismatch between the telephone and the IP 
Telephony gateway phone port can lead to near-end echo. 
The ATA device has a near-end echo canceller that 
compensates for impedance match. The ATA device also 
implements an echo suppressor with comfort noise 
generator (CNG) so that any residual echo is not 
noticeable. Echo Cancellation is configured in the 
Regional, Line, and PSTN Line tabs. See 

“ATA Voice Field 

Reference,” on page 115

.

Feature

Description

Summary of Contents for Linksys SPA3102

Page 1: ...Cisco Small Business SPA2102 SPA3102 SPA8000 SPA8800 PAP2T Analog Telephone Adapters ADMINISTRATION GUIDE ...

Page 2: ...ress Cisco Systems Cisco Systems Capital the Cisco Systems logo Cisco Unity Collaboration Without Limitation EtherFast EtherSwitch Event Center Fast Step Follow Me Browsing FormShare GigaDrive HomeLink Internet Quotient IOS iPhone iQuick Study IronPort the IronPort logo LightStream Linksys MediaTone MeetingPlace MeetingPlace Chime Sound MGX Networkers Networking Academy Network Registrar PCNow PIX...

Page 3: ...y 28 Other ATA Software Features 28 Chapter 2 Basic Administration and Configuration 36 Basic Services and Equipment Required 36 Downloading Firmware 37 Basic Installation and Configuration 37 Upgrading the Firmware for the ATA Device 37 Setting up Your ATA Device 38 Using the Administration Web Server 39 Connecting to the Administration Web Server 40 Setting Up the WAN Configuration for Your ATA ...

Page 4: ...ter s NAT Mechanism 52 Firewalls and SIP 53 Configuring SIP Timer Values 53 Chapter 4 Configuring Voice Services 54 Supported Codecs 54 Using a FAX Machine 55 Fax Troubleshooting 56 Managing Caller ID Service 58 Silence Suppression and Comfort Noise Generation 60 Configuring Dial Plans 61 About Dial Plans 61 Editing Dial Plans 70 Secure Call Implementation 71 Enabling Secure Calls 71 Secure Call D...

Page 5: ...er 88 About the Streaming Audio Server 88 Configuring the Streaming Audio Server 90 Using the IVR with an SAS Line 91 Chapter 6 Configuring the PSTN FXO Gateway on the SPA3102 92 Connecting to PSTN and VoIP Services 92 How VoIP To PSTN Calls Work 93 One Stage Dialing SPA3102 and SPA8800 93 Two Stage Dialing SPA3102 94 How PSTN To VoIP Calls Work 95 Terminating Gateway Calls 96 VoIP Outbound Call R...

Page 6: ... Settings section 108 PPPoE Settings section 108 Optional Settings section 109 MAC Clone Settings section 110 Remote Management section 110 QOS Settings section 110 VLAN Settings section 111 LAN Status page 111 Networking Service section 111 LAN Networking Settings section 112 Static DHCP Lease Settings section 112 Application page 112 Port Forwarding Settings section 113 DMZ Settings section 113 ...

Page 7: ...tion 132 SDP Payload Types section 134 NAT Support Parameters section 135 Trunking Parameters section SPA8000 138 Regional page 139 Call Progress Tones section 140 Distinctive Ring Patterns section 142 Distinctive Call Waiting Tone Patterns section 143 Distinctive Ring CWT Pattern Names section 144 Ring and Call Waiting Tone Spec section 145 Control Timer Values sec section 146 Vertical Service Ac...

Page 8: ...oIP Gateway Setup section SPA8800 184 FXO Timer Values sec section SPA8800 185 PSTN Disconnect Detection section SPA8800 186 International Control section SPA8800 189 Call Forward Speed Dial Supplementary Services and Ring Settings SPA8000 and SPA8800 190 Trunk Group page SPA8000 191 Line Enable section 191 Network Settings section 191 SIP Settings section 192 Subscriber Information section 195 Di...

Page 9: ... Selective Call Forward Settings section 223 Speed Dial Settings section 223 Supplementary Service Settings section 224 Distinctive Ring Settings section 225 Ring Settings section 226 PSTN User page SPA3102 227 PSTN To VoIP Selective Call Forward Settings section 227 PSTN To VoIP Speed Dial Settings section 228 PSTN Ring Thru Line 1 Distinctive Ring Settings section 228 PSTN Ring Thru Line 1 Ring ...

Page 10: ...Cisco Small Business ATA Administration Guide 10 Contents Product Resources 237 Related Documentation 238 ...

Page 11: ...2 Finding Information in PDF Files on page13 Purpose This document provides information that administrators can use to configure and manage Cisco ATAs that are used in conjunction with the SPA9000 Voice System Audience This document is written for the following audience Service providers offering services using LVS products VARs and resellers who need LVS configuration references System administra...

Page 12: ...Boldface May indicate either of the following A user interface element that you need to click select or otherwise act on A literal value to be entered in a field Italic May indicate either of the following A variable that should be replaced with a literal value The name of a page section or field in the user interface Monospaced Font Indicates code samples or system output ...

Page 13: ...or disk drive Perform advanced searches Finding Text in a PDF Follow this procedure to find text in a PDF file STEP 1 Enter your search terms in the Find text box on the toolbar NOTE By default the Find tool is available at the right end of the Acrobat toolbar If the Find tool does not appear choose Edit Find STEP 2 Optionally click the arrow next to the Find text box to refine your search by choo...

Page 14: ...r STEP 2 Choose Edit Search or click the arrow next to the Find box and then choose Open Full Acrobat Search STEP 3 In the Search window complete the following steps a Enter the text that you want to find b Choose All PDF Documents in From the drop down box choose Browse for Location Then choose the location on your computer or local network and click OK c If you want to specify additional search ...

Page 15: ...ministration Guide 15 STEP 4 When the Results appear click to open a folder and then click any link to open the file where the search terms appear For more information about the Find and Search functions see the Adobe Acrobat online help ...

Page 16: ...ne or more standard telephone RJ 11 phone ports using standard analog telephone equipment The ATA device connects to a wide area IP network such as the Internet through a broadband DSL or cable modem or router The ATA can be used with an onsite call control system such as the SPA9000 Voice System or legacy PBX or with an Internet based call control system Figure 1 ATA Deployment without Onsite Cal...

Page 17: ... and business IP Telephony services delivered over broadband or high speed Internet connections An ATA device maintains the state of each call it terminates and makes the proper reaction to user input events such as on off hook or hook flash The ATA devices use the Session Initiation Protocol SIP open standard so there is little or no involvement by a middle man server or media gateway controller ...

Page 18: ...rates how the different ATA devices provide voice connectivity in a VoIP network Product Name FXS Analog Phone FXO PSTN RJ 45 Internet WAN RJ 45 Ethernet LAN Voice Lines Description PAP2T 2 1 2 Voice adapter with two FXS ports SPA2102 2 1 1 2 Voice adapter with router SPA3102 1 1 1 1 1 Voice adapter with router and PSTN connectivity SPA8000 8 1 Maintenance only 8 Voice adapter with support for up ...

Page 19: ...102 and the SPA8800 act as SIP PSTN gateways The WRTP54G router provides ports for analog telephone devices and provides QoS in the form of priority packet queueing NOTE For information about the WRP400 see the WRP400 Administration Guide SPA3102 Broadband router Broadband router SPA8000 PAP2T DSL cable modem WAG54GP2 AG310 WRP400 RTP300 WRTP54G and SPA2102 Ethernet Wireless LAN Fax up to 4 SPA800...

Page 20: ...assigned by the ATA device DHCP server The address and port are translated by the ATA device using Network Address Translation NAT and Port Address Translation PAT The packet is then routed back to the internal network on the ATA device by the local router or the ISP router Problems can occur with calls between phones connected to the ATA device when an outbound proxy or a router with hairpinning ...

Page 21: ...OTE The IVR functions are accessed by connecting an analog telephone to Line 1 For proper operation the service provider should use an Outbound Proxy to forward all voice traffic when the PAP2T is located behind a router If necessary explicit port ranges can be specified for SIP and RTP Line 1 Line 2 Internet IP Router with hairpinning or Broadband modem ITSP ISP PAP2T LAN WAN Ethernet port Admini...

Page 22: ...conflict with a device on the Ethernet port the network address of the device on the LAN port is automatically changed to 192 168 1 0 NOTE The IVR functions are accessed by connecting an analog telephone to Line 1 For proper operation the service provider should use an Outbound Proxy to forward all voice traffic when the SPA2102 is located behind a router If necessary explicit port ranges can be s...

Page 23: ...device on the Ethernet port the network address of the device on the LAN port is automatically changed to 192 168 1 0 NOTE The IVR functions are accessed by connecting an analog telephone to Line 1 For proper operation the service provider should use an Outbound Proxy to forward all voice traffic when the SPA3102 is located behind a router If necessary explicit port ranges can be specified for SIP...

Page 24: ...t port the network address of the device on the AUX port is automatically changed to 192 168 1 0 In the illustration one fax machine is connected to each pair of ports to illustrate that only one T 38 connection is supported by each of the four pairs of RJ 11 ports Up to four fax machines can be connected to the SPA8000 router but they must be distributed as shown Line 1 Line 2 Internet IP Router ...

Page 25: ...d IP packets to devices connected to its AUX port and that configuration is not supported The SPA8000 also can be configured with trunk groups and trunk lines See SIP Trunking and Hunt Groups on the SPA8000 on page 75 SPA8800 Connectivity As shown in the following figure the SPA8800 has four voice modules that each provide 1 FXS port and 1 FXO port 4 FXS RJ 11 RJ 21 ports 4 FXO ports PSTN PSTN PST...

Page 26: ...is a full featured fully programmable phone adapter that can be custom provisioned within a wide range of configuration parameters This section contains a high level overview of features to provide a basic understanding of the feature breadth and capabilities of the ATA device The following sections describe the factors that contribute to voice quality Voice Supported Codecs on page 26 SIP Proxy R...

Page 27: ...er packet This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs G 726 This low complexity codec supports compressed 16 24 32 and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet This codec provides high voice quality G 729a The ITU G 729 voice coding algorithm is used to compress digitized speech Cisco supports...

Page 28: ...NS server to get a list of hosts in the given domain that provides SIP services If an entry exists the DNS server returns an SRV record that contains a list of SIP proxy servers for the domain with their host names priority listening ports and so on The ATA device tries to contact the list of hosts in the order of their stated priority If the ATA device is currently using a lower priority proxy se...

Page 29: ...The ATA device can buffer incoming voice packets to minimize out of order packet arrival This process is known as jitter buffering The jitter buffer size proactively adjusts or adapts in size depending on changing network conditions The ATA device has a Network Jitter Level control setting for each line of service The jitter level determines how aggressively the ATA device tries to shrink the jitt...

Page 30: ... configured in the DTMF Tx Mode parameter found in the Line tabs See the ATA Voice Field Reference on page 115 Call Progress Tone Generation The ATA device has configurable call progress tones Call progress tones are generated locally on the ATA device so an end user is advised of status such as ringback Parameters for each type of tone for instance a dial tone played back to an end user may inclu...

Page 31: ...time allowed for detection of a hook flash using the Hook Flash Timer parameter on the Regional tab of the administration web server See ATA Voice Field Reference on page 115 Configurable Dial Plan with Interdigit Timers The ATA device has three configurable interdigit timers Initial timeout T Signals that the handset is off the hook and that no digit has been pressed yet Long timeout L Signals th...

Page 32: ...rence on page 115 Syslog and Debug Server Records Syslog and Debug Sever Records log more details than Report Generation and Event Logging Using the configuration parameters the ATA device allows you to select which type of activity events should be logged Syslog and Debug Server allow the information captured to be sent to a Syslog Server Syslog and Debug Server Records are configured in the Syst...

Page 33: ... Protocol SRTP can be used to encrypt voice packets SIP over TLS is configured in the SIP Transport parameter configured in the Line tab s See ATA Voice Field Reference on page 115 Media Loopback SPA2102 SPA3102 and PAP2T devices allow service providers to use media loopback to quantitatively and qualitatively measure the voice quality experienced by the end user One device acts as the audio trans...

Page 34: ...les this feature Reg Retry Long Random Delay Random delay range in seconds to add to the Register Retry Long Intvl parameter when retrying a SIP REGISTER after a failure The default is 0 which disables this feature Reg Retry Intvl Cap The maximum value to cap the exponential back off retry delay The exponential back off retry delay starts with the setting found in the Register Retry Intvl paramete...

Page 35: ...its IP address the communication between the SIP proxy and the device is either severed or degraded Whenever an expected SIP response is not received within a programmable amount of time after the corresponding SIP command is sent the DHCP Renewal on Timeout feature automatically causes the device to request a renewal of its IP address If the DHCP server returns the IP address that it originally a...

Page 36: ...g the Firmware for the ATA Device section on page 37 Setting up Your ATA Device section on page 38 Using the Administration Web Server section on page 39 Upgrading Rebooting and Resyncing Your ATA Device section on page 43 Provisioning Your ATA Device section on page 45 Basic Services and Equipment Required To configure your ATA devices you need the following services and equipment An integrated a...

Page 37: ...ware by going to http www cisco com en US products ps10024 tsd_products_support_series_home html and clicking the Download Software link Basic Installation and Configuration See your the Quick Installation Guide and the User Guide for the ATA model that you are installing If you are configuring the complete SPA9000 Voice System also refer to the documentation for the SPA9000 Voice System Upgrading...

Page 38: ... that the correct device information and product number appear Then click Upgrade e A progress message appears while the upgrade is in progress The success window appears when the upgrade is completed The device reboots f Click OK to close the confirmation message g To verify the upgrade point the web browser to the IP address of the ATA device Check the Router Status page The Software Version fie...

Page 39: ...unt name and the Administrator account name cannot be changed When browsing pages with the Administrator account privilege you can switch to User account privilege by clicking the User Login link If the User account password is set the browser prompts for authentication when you click the User Login link From the User account you can switch to the Administrator account by clicking the Admin Login ...

Page 40: ... to determine the Internet WAN IP address NOTE For more information on the IVR menu see your Quick Installation Guide or User Guide for your device or the LVS Administration Guide STEP 3 Direct the browser to the IP address of the ATA device STEP 4 The Router Status page appears By default the page is in Basic User mode Log on to the administrator view by clicking Admin Login near the top right co...

Page 41: ...m the Connection Type drop down menu b In the Static IP Settings section enter the IP address in the Static IP field the subnet mask in the NetMask field and the default gateway IP address in the Gateway field c In the Optional Settings section enter the DNS server address es in the Primary DNS and optional Secondary DNS fields For PPPoE a Select PPPoE from the Connection Type drop down menu This ...

Page 42: ...r Information section Proxy The proxy server for your ITSP account Proxy and Registration section STEP 4 After making any necessary changes click the Submit All Changes button STEP 5 To verify your progress perform the following tasks After the devices reboot click Voice tab Info Scroll down to the Line 1 Status section of the page Verify that the line is registered Use an external phone to place ...

Page 43: ...bscriber Information You can configure security parameters See the Secure Call Implementation on page 71 for further information Dial Plan You can configure a dial plan for a specific line See the Configuring Dial Plans on page 61 for further information Upgrading Rebooting and Resyncing Your ATA Device The administration web server supports upgrading rebooting and resyncing functions through spec...

Page 44: ...do a resync to a profile specified in the URL which can identify either a TFTP HTTP or HTTPS server The syntax of the Resync URL is as follows http spa ip addr admin resync protocol server name port profile pathname NOTE The SPA resyncs only when it is idle If no parameter follows resync the Profile Rule setting from the Provisioning page is used If no protocol is specified TFTP is assumed If no s...

Page 45: ...g the desired profile URL into the device The ATA device accepts profiles in XML format or alternatively in a proprietary binary format which is generated by a profile compiler tool available from Cisco Find the Profiler Compiler for your ATA at http www cisco com web partners sell smb products voice_and_conferencing html vc_technical_resources The ATA device supports up to 256 bit symmetric key e...

Page 46: ...ironment Requests for SPC tools compiled on other platforms are evaluated on a case by case basis Please contact your sales representative for further information about obtaining the SPC tool The syntax of the plain text file accepted by the profile compiler is a series of parameter value pairs with the value in double quotes Each parameter value pair is followed by a semicolon Here is an example ...

Page 47: ... is a function that allows multiple devices to share the same public routable IP address to establish connections over the Internet NAT is present in many broadband access devices to translate public and private IP addresses To enable VoIP to co exist with NAT some form of NAT traversal is required Some ITSPs provide NAT traversal but some do not If your ITSP does not provide NAT traversal you hav...

Page 48: ...ith a SIP ALG can be used By using a SIP ALG router you have more choices in selecting an ITSP Configuring NAT Mapping with a Static IP Address If the ITSP network does not provide a Session Border Controller functionality and if other requirements are met you can configure NAT mapping to ensure interoperability with the ITSP Requirements You must have an external public IP address that is static ...

Page 49: ...stitute VIA Addr yes Handle VIA rport Insert VIA rport Send Resp To Src Port yes EXT IP Enter the public IP address for your router Voice tab SIP NAT Support Parameters STEP 4 Click Voice tab Line N where N represents the line interface number STEP 5 Scroll down to the NAT Settings section NAT Mapping Enable Choose YES NAT Keep Alive Enable Choose YES optional Voice tab Line N NAT Settings STEP 6 ...

Page 50: ...T Mechanism on page 52 You must have a computer running STUN server software The LAN switch must be configured to enable Spanning Tree Protocol and Port Fast on the ports to which the SPA devices are connected NOTE Use NAT mapping only if the ITSP network does not provide a Session Border Controller functionality STEP 1 Connect to the administration web server and choose Admin access with Advanced...

Page 51: ... STEP 5 Scroll down to the NAT Settings section NAT Mapping Enable Choose yes NAT Keep Alive Enable Choose yes optional Voice tab Line N NAT Settings NOTE Your ITSP may require the SPA device to send NAT keep alive messages to keep the NAT ports open permanently Check with your ITSP to determine the requirements STEP 6 Click Submit All Changes NOTE You also need to configure the firewall settings ...

Page 52: ... a syslog server is configured and is ready to receive syslog messages STEP 1 Make sure you do not have firewall running on your PC that could block the syslog port port 514 by default STEP 2 Connect to the administration web server and choose Admin access with Advanced settings STEP 3 To enable debugging complete the following tasks a Click Voice tab System b In the Debug Server field enter the I...

Page 53: ...the following ports are not blocked SIP ports UDP port 5060 through 5063 which are used for the ITSP line interfaces RTP ports 16384 to 16482 Also disable SPI Stateful Packet Inspection if this function exists on your firewall Configuring SIP Timer Values The default timer values should be adequate in most circumstances However you can adjust the SIP timer values as needed to ensure interoperabili...

Page 54: ...figuring Dial Plans on page 61 Secure Call Implementation on page 71 SIP Trunking and Hunt Groups on the SPA8000 on page 75 Supported Codecs The following list shows the current supported codecs for each ATA device If you need to change the G711u codec which is configured by default set your preferred codecs in the FXS Line tab s Audio Configuration You may set your first second and third preferre...

Page 55: ...tion while the SPA8000 and SPA8800 support one connection for each pair of FXS ports 1 2 3 4 5 6 and 7 8 for a maximum of four connections Follow this procedure to optimize fax completion rates STEP 1 Upgrade the ATA firmware to the latest version STEP 2 Ensure that you have enough bandwidth for uplink and downlink For G 711 fallback it is recommend to have approximately 100Kbps For T 38 allocate ...

Page 56: ...n the Line tab for the FXS port to which the FAX machine is connected FAX_Passthru_Method ReINVITE NOTE If a T 38 call cannot be set up then the call should automatically revert to G 711 fallback STEP 6 If you are using a Cisco media gateway use the following settings Make sure the Cisco gateway is correctly configured for T 38 with the SPA dial peer For example fax protocol T38 fax rate voice fax...

Page 57: ...eb server page You can send this configuration file to Technical Support STEP 6 Enable and capture the debug log For instructions refer to Appendix C Troubleshooting NOTE You may also capture data using a sniffer trace STEP 7 Identify the type of fax machine connected to the ATA device STEP 8 Contact technical support If you are an end user of VoIP products contact the reseller or Internet telepho...

Page 58: ...y reversal and before first ring ETSI DTMF With PR CID only DTMF sent after polarity reversal and DTAS and before first ring ETSI DTMF After Ring CID only DTMF sent after first ring no polarity reversal or DTAS ETSI FSK CID CIDCW and VMWI FSK sent after DTAS but no polarity reversal and before first ring Waits for ACK from CPE after DTAS for CIDCW ETSI FSK With PR UK CID CIDCW and VMWI FSK is sent...

Page 59: ...ed CID methods Bellcore ETSI FSK and ETSI FSK With PR Off Hook Caller ID This is used to delivery caller id on incoming calls when the attached phone is off hook see the following figure This can be call waiting caller ID CIDCW or to notify the user that the far end party identity has changed or updated such as due to a call transfer This is available only for FSK based CID methods Bellcore ETSI F...

Page 60: ...tion is activated This is accomplished by removing and not transmitting the natural silence that occurs in normal two way connection The IP bandwidth is used only when someone is speaking During the silent periods of a telephone call additional bandwidth is available for other voice calls or data traffic because the silence packets are not being transmitted across the network Comfort Noise Generat...

Page 61: ...ing Dial Plans on page 70 About Dial Plans This section provides information to help you understand how dial plans are implemented Refer to the following topics Digit Sequences on page 61 Digit Sequence Examples on page 64 Acceptance and Transmission the Dialed Digits on page 66 Dial Plan Timer Off Hook Timer on page 67 Interdigit Long Timer Incomplete Entry Timer on page 68 Interdigit Short Timer...

Page 62: ...ser to press 3 5 6 7 8 or period Enter a period for element repetition The dial plan accepts 0 or more entries of the digit For example 01 allows users to enter 0 01 011 0111 and so on dialed substituted For sequence substitution use this format to indicate that certain dialed digits are replaced by other characters when the sequence is transmitted The dialed digits can be zero or more characters ...

Page 63: ...til the user presses 1 exclamation point For number barring enter an exclamation point to prohibit a dial sequence pattern EXAMPLE 1900xxxxxxx The system rejects any 11 digit sequence that begins with 1900 xx Enter an asterisk to allow the user to enter a 2 digit star code S0 or L0 For Interdigit Timer Master Override enter S0 to reduce the short inter digit timer to 0 seconds or enter L0 to reduc...

Page 64: ...he following string 1 8 xxx Local dialing with seven digit number EXAMPLE 1 8 xx 9 xxxxxxx 9 1 2 9 xxxxxxxxx 8 1212 xxxxxxx 9 1 2 9 xxxxxxxxx 9 1 900 xxxxxxx 9 011xxxxxx 0 49 111 9 xxxxxxx After a user presses 9 an external dial tone sounds The user can enter any seven digit number as in a local call Local dialing with 3 digit area code and a 7 digit local number EXAMPLE 1 8 xx 9 xxxxxxx 9 1 2 9 x...

Page 65: ...xxx 9 1 900 xxxxxxx 9 011xxxxxx 0 49 11 9 1 900 xxxxxxx This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content such as 1 900 numbers in the U S After the user press 9 an external dial tone sounds If the user enters an 11 digit number that starts with the digits 1900 the call is rejected U S international dialing ...

Page 66: ...number is rejected The dialed digits exactly match one sequence in the dial plan If the sequence is allowed by the dial plan the number is accepted and is transmitted according to the dial plan If the sequence is blocked by the dial plan the number is rejected A timeout occurs The number is rejected if the dialed digits are not matched to a digit sequence in the dial plan within the time specified...

Page 67: ... user hears a reorder fast busy tone after the specified number of seconds Examples for the Dial Plan Timer Allow more time for users to start dialing after taking a phone off hook EXAMPLE P9 9 8 1408 2 9 xxxxxx 9 8 1 2 9 xxxxxxxxx 9 8 011xx 9 8 xx 1 8 xx P9 After taking a phone off hook a user has 9 seconds to begin dialing If no digits are pressed within 9 seconds the user hears a reorder fast b...

Page 68: ... of seconds if no number is entered after L the default timer of 5 seconds applies Note that the timer sequence appears to the left of the initial parenthesis for the dial plan Example for the Interdigit Long Timer EXAMPLE L 15 9 8 1408 2 9 xxxxxx 9 8 1 2 9 xxxxxxxxx 9 8 011xx 9 8 xx 1 8 xx L 15 This dial plan allows the user to pause for up to 15 seconds between digits before the Interdigit Long ...

Page 69: ...e timer for the entire dial plan EXAMPLE S 6 9 8 1408 2 9 xxxxxx 9 8 1 2 9 xxxxxxxxx 9 8 011xx 9 8 xx 1 8 xx S 6 While entering a number with the phone off hook a user can pause for up to 15 seconds between digits before the Interdigit Short Timer expires This setting is especially helpful to users such as sales people who are reading the numbers from business cards and other printed materials whi...

Page 70: ...ace number STEP 3 Scroll down to the Dial Plan section STEP 4 Enter the digit sequences in the Dial Plan field For more information see About Dial Plans on page 61 STEP 5 Click Submit All Changes Resetting the Control Timers You can use the following procedure to reset the default timer settings for all calls NOTE If you need to edit a timer setting only for a particular digit sequence or type of ...

Page 71: ...The information is transported by base64 encoding embedded in the message body of SIP INFO requests and responses using a proprietary format If the second stage is successful the ATA device plays a special Secure Call Indication Tone for a short time to indicate to both parties that the call is secured and that RTP traffic in both directions is being encrypted If the user has a phone that supports...

Page 72: ... check after receiving the Mini Certificate of the called party Secure Call Details Looking at the second stage of setting up a secure call in greater detail this stage can be further divided into two steps STEP 1 The caller sends a Caller Hello message base64 encoded and embedded in the message body of a SIP INFO request to the called party with the following information Message ID 4B Version and...

Page 73: ...st also be provisioned for each subscriber The CA public key is used to verify the MC received from the other end If the MC is invalid the call will not switch to secure mode The MC and the 1024 bit CA public key are concatenated and base64 encoded into the single parameter Mini Certificate The 512 bit private key is base64 encoded into the SRTP Private Key parameter which should be kept secret li...

Page 74: ... used in the INVITE when making the call such as 14083331234 The maximum length is 16 characters expire date is the expiration date of the MC such as 00 00 00 1 1 34 34 2034 Internally the date is encoded as a fixed 12B string 000000010134 The tool generates the Mini Certificate and SRTP Private Key parameters that can be provisioned EXAMPLE gen_mc ca_key Joe Smith 14085551234 00 00 00 1 1 34 This...

Page 75: ...der For outbound calls SIP Trunking ensures that all calls on a trunk line can be identified by the trunk number and a common caller ID This feature helps you to ensure that calls are directed to available lines and that work groups such as sales teams can work together to answer calls In addition teams can project a common identity when placing outbound calls on a trunk This section provides info...

Page 76: ...an be configured on each trunk group with a distinct phone number Each of the eight SPA8000 lines can be configured either as a standalone line as in a classic ATA FXS port or as a trunk line that is associated with a trunk group Inbound calling A trunk group offers a single number for callers to call into the small business with the capability to programmatically ring one or more trunk lines Outb...

Page 77: ...hanges signaling directly with the ITSP equipment As a trunk line the Line UA exchanges signaling with the internal proxy server only The Internal Proxy Server handles all SIP signalling between both ends of the call from call establishment to termination RTP Path Whether the line is standalone or a member of a trunk group the Line UA exchanges RTP packets directly with the ITSP equipment Phone 1 ...

Page 78: ...ches full capacity it will not attempt to failover to use other trunks You can configure this setting in the Voice tab Trunk T1 T4 page Subscriber Information section Call Capacity field For more information see Configuring a Trunk Group on page 82 Inbound Call Routing for a Trunk Group An incoming call is handled as follows STEP 1 When an incoming call is detected by the Trunk UA the UA first che...

Page 79: ...o a trunk group through the Voice tab Line page Trunk Group field The Trunk UA does not ring any standalone lines that are included in the Contact List The Trunk UA rings any trunk line that is included in the list even if it is not assigned to the particular trunk group for this Contact List You can instruct the SPA8000 to hunt only the phones that are on hook through the Voice tab SIP page Trunk...

Page 80: ... until the caller either hangs up or the call is answered Exceptions This value is ignored if algo all or interval but it must be present and should be set to 1 cfwd target If the call is unanswered and the maximum hunting duration has been met the call is forwarded to the specified number When forwarding the call the SPA8000 sends a 302 response to the ITSP NOTE The call forward settings for the ...

Page 81: ...he PBX finds an open line it takes the line off hook and bridges the audio between the PBX phone and the line On detecting the off hook signal the SPA8000 Line UA plays dial tone and ready to collect digits from the PBX phone STEP 2 As the PBX phone user dials the number the Line UA applies its dial plan to the number If the Line UA detects an invalid number it rejects the all by playing reorder t...

Page 82: ...want to associate with each trunk group that you are configuring Refer to the following example STEP 1 Connect to the administration web server and choose Admin access with Advanced settings STEP 2 Assign each line to a trunk group as needed a Click Voice tab Ln where n represents the number of the line interface b In the Trunk Group field near the top of the line configuration page choose a trunk...

Page 83: ...unt c In the Call Capacity field enter the maximum number of concurrent calls allowed by your ITSP or leave the default setting unlimited 16 calls d In the Contact List field modify the contact list as needed See Contact List for a Trunk Group on page 79 e Repeat this step for each trunk group that you need to configure STEP 4 Click Submit All Changes Trunk Group Management You can check the statu...

Page 84: ...r The page shows the following information External The called number Station The SPA8000 line that is in use for this call Direction The direction of the call either Outbound or Inbound State The state of the call Calling An outbound call was initiated but is not ringing at the other end Proceeding The outbound call is ringing at the other end Ringing An inbound call is ringing Connected The call...

Page 85: ... section includes information about other topics that may be of interest when you are configuring trunk groups Voice mail There is no individual mail box for a trunk line For example if lines 1 2 3 and 4 belong the trunk group T1 then the four lines implicitly share the same voice mail box from the ITSP When there is new voice mail waiting in the trunk mail box the UAs for all four lines will be n...

Page 86: ...available It plays an internally stored music file repeatedly The unit ships with a default music file Romance de Amor You can override this file by downloading a new file into the unit by using TFTP Refer to the following topics Using the Internal Music Source on page 86 Changing the Music File for the Internal Music Source on page 87 Using the Internal Music Source To use the internal music sour...

Page 87: ...e for the Internal Music Source The following resources are required to change the music file for the internal music source TFTP server software The IP address of the administration computer that is connected to the SPA9000 A music source in G 711u format sampled at 8000 samples sec up to 65 5 seconds in length with no header information STEP 1 Before you begin make sure that you have TFTP server ...

Page 88: ...and configure a streaming audio server SAS It includes the following topics About the Streaming Audio Server on page 88 Configuring the Streaming Audio Server on page 90 Using the IVR with an SAS Line on page 91 About the Streaming Audio Server The Streaming Audio Server SAS feature lets you attach an audio source to an FXS port and use it as a streaming audio source device If the unit has multipl...

Page 89: ... If an incoming call is auto answered but later the FXS port changes to on hook the call is not terminated but continues to stream silence packets to the caller The SAS line can be set up to refresh each streaming audio session periodically using a SIP re INVITE message which detects if the connection to the caller is down If the caller does not respond to the refresh message the SAS line terminat...

Page 90: ...er the following settings Display Name Enter an extension number of name for the FXS 1 port such as Receptionist Area Fax Machine User ID Enter a three to four digit extension number that is not is use by another extension c In the Streaming Audio Server SAS section choose yes from the SAS Enable drop down list STEP 4 Click Submit All Changes STEP 5 Configure each phone to use this audio source as...

Page 91: ...er needs to follow the following steps STEP 1 Power off the ATA device STEP 2 Connect a phone to the port and make sure the phone is on hook STEP 3 Power on the ATA device STEP 4 Pick up handset and press to invoke IVR in the usual way If the ATA device boots and finds that the SAS line is on hook it does not remove battery from the line so that IVR may be used But if the ATA device boots up and f...

Page 92: ... on page100 Connecting to PSTN and VoIP Services Both the SPA3102 and the SPA8800 allow your analog and IP phones to participate in calls over the VoIP network and the PSTN Both devices provide a VoIP To PSTN calling function which is referred to as a PSTN gateway as well as a PSTN To VoIP calling function which is referred to as a VoIP gateway These ATAs function somewhat differently because they...

Page 93: ...8800 the VoIP caller establishes a connection with the PSTN Line by way of a standard SIP INVITE request addressed to the PSTN Line One Stage Dialing SPA3102 and SPA8800 One stage dialing allows a call to be started over VoIP and then immediately get a dial tone on the PSTN When you take a phone off hook and dial a number the call is automatically routed to the VoIP or the PSTN based on the dial p...

Page 94: ...ller is prompted to enter a PIN number after the SPA3102 answers the call The PIN number must end with a key The inter PIN digit timeout is 10 seconds not configurable Up to eight VoIP caller PIN numbers can be configured on the SPA3102 A dial plan can be selected for each PIN number If the caller enters a wrong PIN or the SPA3102 times out waiting for more PIN digits the SPA3102 tears down the ca...

Page 95: ... The PSTN caller enters the target telephone number The collected digits are processed by the default dial plan On the SPA3102 you can add PIN authentication to the basic flow 1 When a PSTN call comes in to the ATA device and is unanswered after a configurable number of rings then the ATA device takes the FXO port off hook 2 The SPA3102 prompts the caller to enter the PIN number followed by the ke...

Page 96: ...ne is disconnected from the PSTN service or if the PSTN switch provides a CPC signal A polarity reversal or disconnect tone is detected at the FXO port There is no voice activity for a configurable period of time in either direction at the FXO port When any of the above conditions occur the ATA device takes the FXO port on hook and sends a BYE request to end the VoIP call leg On the other hand whe...

Page 97: ...making certain calls In general you can specify any gateway address in the dial plan In addition three parameters are added that can be used with call routing usr User id used for authentication with the given gateway pwd Password used for authentication with the given gateway nat Enable or disable NAT mapping when calling the gateway The following table lists some examples Example Description 9 x...

Page 98: ...ally route all outbound calls to the internal gateway using the parameter listed below For SPA3102 you can configure this setting on the Voice tab Line page For SPA8800 you can configure this setting on the Voice tab Phone page For more information see VoIP Fallback to PSTN section SPA3102 and SPA8800 on page180 Sharing One VoIP Account Between the FXS and PSTN Lines SPA3102 On the SPA3102 both th...

Page 99: ...u can forward all callers with 408 area code to 14081234567 or all callers with 800 area code to 18005558355 This is the number for Tell Me When this syntax is used authentication is not used and the target PSTN number is automatically dialed by the ATA device after the caller is forwarded to gw0 Other Options This section describes other options provided by the SPA3102 and the SPA8800 PSTN Call t...

Page 100: ...arios This section describes some typical scenarios where the ATA device can be applied Some terms are introduced in the first few sections and reused in later sections This section includes the following topics PSTN to VoIP Call with and Without Ring Thru section on page101 VoIP to PSTN Call With and Without Authentication section on page 101 Call Forwarding to PSTN Gateway SPA3102 and SPA8800 se...

Page 101: ...t entering the PIN In this case the default PSTN dial plan is also used The same scenario can be implemented using Ring Thru When the PSTN line rings Line 1 rings also This feature is called Ring Thru If Line1 is picked up before the VoIP gateway auto answers it is connected to the PSTN call Line 1 hears a call waiting tone if it is already connected to another call VoIP to PSTN Call With and With...

Page 102: ...call with a 403 This rule applies regardless of the authentication method even when the source IP address of the INVITE request is in the VoIP Access List Using HTTP Digest Authentication SPA3102 The same scenario can be implemented with HTTP digest authentication when the calling device supports the configuration of a auth ID and password to access the ATA device PSTN gateway When the VoIP caller...

Page 103: ... the PSTN gateway It includes the following topics Forward On No Answer to the PSTN Gateway section on page103 Forward All to the PSTN gateway section on page 104 Forward to a Particular PSTN Number section on page104 Forward On Busy to PSTN Gateway or Number section on page 104 On the SPA3102 you can configure Call Forward settings on the User page On SPA8800 the same parameters are set on the Ph...

Page 104: ... the call the call forward rule is ignored and Line 1 continues to ring Forward to a Particular PSTN Number In this scenario the forward destination is set to target number gw0 This is the same as in the previous examples except that the ATA device automatically dials the given target number on the PSTN line right after it answers the VoIP call leg This is a special case of one stage dialing where...

Page 105: ...guide for the router After you click the Router tab on the SPA2102 SPA3102 or the Network tab on the SPA8000 and SPA8800 you can choose the following pages Router Status page on page105 WAN Status page on page 107 LAN Status page on page111 Application page on page112 NOTE Not all fields listed may be applicable to your ATA device or your setup Router Status page You can use the Router tab Status ...

Page 106: ...tes a new RC unit that is ready for provisioning If the unit has already retrieved its customized profile this field displays the name of the company that provisioned the unit Current Time Current date and time of the system for example 10 3 2003 16 43 00 Elapsed Time Total time elapsed since the last reboot of the system for example 25 days and 18 12 36 WAN Connection Type The connection type DHC...

Page 107: ...page111 Router tab WAN Setup page Internet Connection Settings section Broadcast Pkts Sent Total number of broadcast packets sent Broadcast Bytes Sent Total number of broadcast packets received Broadcast Pkts Recv Total number of broadcast bytes sent Broadcast Bytes Recv Total number of broadcast bytes received and processed Broadcast Pkts Dropped Total number of broadcast packets received but not...

Page 108: ... by ATA device when DHCP is disabled The default is 255 255 255 0 Gateway The default gateway used by ATA device when DHCP is disabled The default is 0 0 0 0 PPPoE Login Name The account name assigned by the ISP for connecting on a Point to Point Protocol over Ethernet PPPoE link PPPoE Login Password The password assigned by the ISP for connecting on a Point to Point Protocol over Ethernet PPPoE l...

Page 109: ... is enabled you can enter the IP address of a primary or secondary DNS server in addition to DHCP supplied DNS servers When DHCP is disabled enter the primary and secondary DNS server The default is 0 0 0 0 DNS Service Order The method for selecting the DNS server Manual enter the IP address of the DNS server manually that is do not look at the DHCP supplied DNS table Manual DHCP and DHCP Manual D...

Page 110: ...n Use Quality of Service QoS to assign different priority levels to different types of data transmissions Enable MAC Clone Service To use MAC Address cloning select Yes Default is No Cloned MAC Address Use when your ISP requires a certain MAC address It s usually the address for your PC Enable WAN Web Server Allows or prevents access to the administration web server from a computer that is not dir...

Page 111: ...Settings section on page112 Router tab LAN Setup page Networking Service section Maximum Uplink Speed The maximum bandwidth for LAN to WAN throughput The default is 128 kbps Enable VLAN Allows yes or prevents no VLAN access NOTE Choose yes if your ATA device is connected to a switch that uses VLAN tagging VLAN ID The VLAN tag for the VLAN to which the ATA device is assigned Networking Service NAT ...

Page 112: ...Port Forwarding Settings section on page113 DMZ Settings section on page113 Miscellaneous Settings section on page114 LAN IP Address IP address of the ATA device on the LAN side LAN Subnet Mask IP address for subnet mask Enable DHCP Server The status of the DHCP server where Yes is enabled and No is disabled DHCP Lease Time Provided by the DHCP Server IP renewal process begins when the time expire...

Page 113: ...more secure because it only opens the ports you want to have opened while DMZ hosting opens all the ports of one computer exposing the computer to the Internet Enable Options are Yes or No Service Name Any name to call the port forwarding starting port Starting Port The starting port of the port range you wish to forward Ending Port The ending port of the port range you wish to forward Protocol Th...

Page 114: ...ion Multicast Passthru Used for passing multicast traffic Options are disabled inbound outbound inbound and outbound Starting Port A port identified as a reserve port and that is not used for NAT translation That is if there is a conflict if port forwarding is set on the same port then the port forwarding is cancelled Default is 50000 Num of Ports Reserved Total number of ports reserved Options ar...

Page 115: ...e Provisioning tab see the SPA Provisioning Guide After you click the Voice tab you can choose the following pages Info page on page116 System page on page124 SIP page on page 127 Regional page on page139 Line page on page 160 Trunk Group page SPA8000 on page 191 PSTN Line page SPA3102 on page199 User page on page 221 PSTN User page SPA3102 on page 227 NOTE Not all fields listed may be applicable ...

Page 116: ...The fields on the Info page are read only and cannot be edited Voice tab Info page Product Information section Product Name Model number name Serial Number Serial number Software Version Software version number Hardware Version Hardware version number MAC Address MAC address Client Certificate Status of the client certificate which can indicate if the ATA device has been authorized by your ITSP Cu...

Page 117: ... redundant packets RTP Bytes Recv Total number of RTP bytes received SIP Messages Sent Total number of SIP messages sent including retransmissions SIP Bytes Sent Total number of bytes of SIP messages sent including retransmissions SIP Messages Recv Total number of SIP messages received including retransmissions SIP Bytes Recv Total number of bytes of SIP messages received including retransmissions...

Page 118: ...mber of the last caller Mapped SIP Port Port number of the SIP port mapped by NAT Call 1 and 2 State May take one of the following values Idle Collecting PSTN Pin Invalid PSTN PIN PSTN Caller Accepted Connected to PSTN Call 1 and 2 Tone Type of tone used by the call Call 1 and 2 Encoder Codec used for encoding Call 1 and 2 Decoder Codec used for decoding Call 1 and 2 FAX Status of the fax pass thr...

Page 119: ...d 2 Packets Recv Number of packets received Call 1 and 2 Bytes Sent Number of bytes sent Call 1 and 2 Bytes Recv Number of bytes received Call 1 and 2 Decode Latency Number of milliseconds for decoder latency Call 1 and 2 Jitter Number of milliseconds for receiver jitter Call 1 and 2 Round Trip Delay Number of milliseconds for delay Call 1 and 2 Packets Lost Number of packets lost Call 1 and 2 Pac...

Page 120: ...DNS server assigned to the ATA device Secondary DNS Displays the secondary DNS server assigned to the ATA device PSTN Hook State Hook state of the FXO port Either On or Off PSTN Line Voltage The voltage existing on the PSTN line PSTN Loop Current The current milliamperes existing on the local loop Registration State Indicates if the line has registered with the SIP proxy Last Registration At Last ...

Page 121: ...Gateway Call Timeout PSTN Activity Timer Shows the time ms before the SPA disconnects the current gateway unless the PSTN side has some audio activity Mapped SIP Port Port number of the SIP port mapped by NAT Call Type May take one of the following values PSTN Gateway Call VoIP To PSTN Call VoIP Gateway Call PSTN To VoIP Call PSTN To Line 1 PSTN call ring through and answered by Line 1 Line 1 Forw...

Page 122: ... of the party at the VoIP call leg PSTN Peer Name Name of the party at the PSTN call leg VoIP Peer Number Phone number of the party at the VoIP call leg PSTN Peer Number Phone number of the party at the PSTN call leg VoIP Call Encoder Audio encoder being used for the VoIP call leg VoIP Call Decoder Audio decoder being used for the VoIP call leg VoIP Call FAX Status of the fax pass through mode VoI...

Page 123: ...al Time Protocol traffic for Call 1 2 Registration State Indicates if the line has registered with the SIP proxy Last Registration At Last date and time the line was registered Next Registration In Number of seconds before the next registration renewal Message Waiting Indicates whether you have new voice mail waiting Options are either Yes or No This value is updated when voice mail notification i...

Page 124: ...em page System Configuration section Restricted Access Domains This feature is used when implementing software customization Enable Web Server Enable disable web server of the ATA device This feature should only be used on firmware version 1 0 9 or later The default is yes This field is only found in the PAP2T Web Server Port Port number of the ATA device administration web server The default is 8...

Page 125: ...efault is 255 255 255 0 Gateway The default gateway used by ATA device when DHCP is disabled The default is 0 0 0 0 Host Name The host name of the ATA device Domain The network domain of the ATA device Primary DNS DNS server used by ATA device in addition to DHCP supplied DNS servers if DHCP is enabled when DHCP is disabled this is the primary DNS server The default is 0 0 0 0 Secondary DNS Sets t...

Page 126: ...s the server for logging ATA device system information and critical events If both Debug Server and Syslog Server are specified Syslog messages are also logged to the Debug Server Debug Server The debug server name and port This feature specifies the server for logging ATA device debug information The level of detailed output depends on the debug level parameter setting Debug Level The higher the ...

Page 127: ...134 NAT Support Parameters section section on page135 Trunking Parameters section SPA8000 section on page138 Voice tab SIP page SIP Parameters section Debug Level Determines the level of debug information that is generated Select 0 1 2 or 3 from the drop down menu The higher the debug level the more debug information is generated The default is 0 which indicates that no debug information is genera...

Page 128: ...sage to signal a DTMF event The default is application dtmf relay Hook Flash MIME Type MIME Type used in a SIP INFO message to signal a hook flash event The default is application hook flash 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 no Use Compact Header Lets you use compact S...

Page 129: ...ax 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 ...

Page 130: ...r value If you enter 0 the Expires header is not included in the request The default is 240 Range 0 231 1 ReINVITE Expires ReINVITE request Expires header value If you enter 0 the Expires header is not included in the request The default is 30 Range 0 231 1 Reg Min Expires Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter If the prox...

Page 131: ... The default is 0 which disables this feature Reg Retry Intvl Cap The maximum value to cap the exponential back off retry delay which starts at Register Retry Intvl and doubles on every REGISTER retry after a failure In other words the retry interval is always at Register Retry Intvl seconds after a failure If this feature is enabled Reg Retry Random Delay is added on top of the exponential back o...

Page 132: ...rt number for RTP transmission and reception The RTP Port Min and RTP Port Max parameters should define a range that contains at least 4 even number ports such as 100 106 The default is 16384 RTP Port Max Maximum port number for RTP transmission and reception The default is 16482 RTP Packet Size Packet size in seconds which can range from 0 01 to 0 16 Valid values must be a multiple of 0 01 second...

Page 133: ... platform software version such as Cisco ATA device 1 0 31 b The NTP timestamp used in the SR is a snapshot of the ATA device s local time not the time reported by an NTP server If the ATA device receives a RR from the peer it attempts to compute the round trip delay and show it as the Call Round Trip Delay value ms in the Info section of ATA device web page The default is 0 No UDP Checksum Select...

Page 134: ...namic Payload G 726 24 dynamic payload type The valid range is 96 127 The default is 97 G726r40 Dynamic Payload G 726 40 dynamic payload type The valid range is 96 127 The default is 96 G729b Dynamic Payload G 729b dynamic payload type The valid range is 96 127 The default is 99 NSE Codec Name NSE codec name used in SDP The default is NSE AVT Codec Name AVT codec name used in SDP The default is te...

Page 135: ...P Handle VIA received If you select yes the ATA device processes the received parameter in the VIA header this value is inserted by the server in a response to anyone of its requests If you select no the parameter is ignored Select yes or no from the drop down menu The default is no Handle VIA rport If you select yes the ATA device processes the rport parameter in the VIA header this value is inse...

Page 136: ...bled and a valid STUN server is available the ATA device can perform a NAT type discovery operation when it powers on It contacts the configured STUN server and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests If the ATA device detects symmetric NAT or a symmetric firewall NAT mapping is disabled The default is no STUN Server IP address or fully quali...

Page 137: ...rnal port mapping number of the RTP Port Min number If this value is not zero the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range The default is 0 NAT Keep Alive Intvl Interval between NAT mapping keep alive messages The default is 15 ...

Page 138: ...onses 1 line excl NTFY Logs the start line only for all messages except NOTIFY requests responses 1 line excl REG Logs the start line only for all messages except REGISTER requests responses 1 line excl OPT NTFY REG Logs the start line only for all messages except OPTIONS NOTIFY and REGISTER requests responses full Logs all SIP messages in full text full excl OPT Logs all SIP messages in full text...

Page 139: ... section section on page145 Control Timer Values sec section section on page146 Vertical Service Activation Codes section section on page 148 Vertical Service Announcement Codes section SPA2102 SPA8000 section on page 154 Outbound Call Codec Selection Codes section section on page154 Miscellaneous section section on page156 Hunt Policy This parameter can be used to modify the hunting behavior for ...

Page 140: ...encountered in the dial plan The default is 420 19 10 0 1 Prompt Tone Prompts the user to enter a call forwarding phone number The default is 520 19 620 19 10 0 1 2 Busy Tone Played when a 486 RSC is received for an outbound call The default is 480 19 620 19 10 5 5 1 2 Reorder Tone Played when an outbound call has failed or after the far end hangs up during an established call Reorder Tone is play...

Page 141: ...Tone Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call The RSC to trigger this tone is configurable on the SIP screen The default is 914 16 1371 16 1777 16 20 274 0 1 274 0 2 380 0 3 0 4 0 SIT3 Tone Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call The RSC to trigger this tone is configurable on the SIP scree...

Page 142: ...97 19 507 19 15 0 2 0 2 1 1 1 2 1 2 VoIP PIN Tone Specification of the tone played to prompt a VoIP caller for a PIN number if PIN authentication is selected and the caller requires authentication to use the PSTN gateway This setting applies to the SPA3102 only The default is 600 10 0 1 1 1 1 1 1 1 1 1 5 1 PSTN PIN Tone Specification of the tone played to prompt a PSTN caller for a PIN number if P...

Page 143: ...2 4 Ring7 Cadence Cadence script for distinctive ring 7 The default is 60 4 2 4 2 4 4 Ring8 Cadence Cadence script for distinctive ring 8 The default is 60 0 25 9 75 Ring9 Cadence Cadence script for distinctive ring 9 This field is for the SPA2102 and SPA8000 only The default is 60 4 2 4 2 CWT1 Cadence Cadence script for distinctive CWT 1 The default is 30 3 9 7 CWT2 Cadence Cadence script for dis...

Page 144: ... Alert Info Header to pick distinctive ring CWT 1 for the inbound call The default is Bellcore r1 Ring2 Name Name in an INVITE s Alert Info Header to pick distinctive ring CWT 2 for the inbound call The default is Bellcore r2 Ring3 Name Name in an INVITE s Alert Info Header to pick distinctive ring CWT 3 for the inbound call The default is Bellcore r3 Ring4 Name Name in an INVITE s Alert Info Head...

Page 145: ...ng Frequency 25 Ring Voltage 80V Ring7 Name Name in an INVITE s Alert Info Header to pick distinctive ring CWT 7 for the inbound call The default is Bellcore r7 Ring8 Name Name in an INVITE s Alert Info Header to pick distinctive ring CWT 8 for the inbound call The default is Bellcore r8 Ring9 Name Name in an INVITE s Alert Info Header to pick distinctive ring CWT 9 for the inbound call This field...

Page 146: ... hook event is ignored Range 0 1 0 4 seconds The default is 0 1 Hook Flash Timer Max Maximum on hook time before off hook qualifies as hook flash More than this the on hook event is treated as on hook no hook flash event Range 0 4 1 6 seconds The default is 0 9 Callee On Hook Delay Phone must be on hook for at this time in sec before the ATA device will tear down the current inbound call It does n...

Page 147: ... one digit if at least one matching sequence is complete as dialed but more dialed digits would match other as yet incomplete sequences Range 0 64 seconds The default is 3 CPC Delay Delay in seconds after caller hangs up when the ATA device starts removing the tip and ring voltage to the attached equipment of the called party Range 0 255 seconds ATA device has had polarity reversal feature since r...

Page 148: ...0 Range 0 to 1 000 second Resolution is 0 001 second The default is 0 CPC disabled Call Return Code This code calls the last caller The default is 69 Call Redial Code Redials the last number called This field is not found in the PAP2T The default is 07 Blind Transfer Code Begins a blind transfer of the current call to the extension specified after the activation code The default is 98 Call Back Ac...

Page 149: ...ast Act Code Forwards the last inbound or outbound calls to the extension specified after the activation code The default is 63 Cfwd Last Deact Code Cancels call forwarding of the last inbound or outbound calls The default is 83 Block Last Act Code Blocks the last inbound call The default is 60 Block Last Deact Code Cancels blocking of the last inbound call The default is 80 Accept Last Act Code A...

Page 150: ...nd call The default is 81 Block CID Per Call Deact Code Removes caller ID blocking on the next inbound call The default is 82 Block ANC Act Code Blocks all anonymous calls The default is 77 Block ANC Deact Code Removes blocking of all anonymous calls The default is 87 DND Act Code Enables the do not disturb feature The default is 78 DND Deact Code Disables the do not disturb feature The default is...

Page 151: ...nd calls are secure by default The default is 18 Secure One Call Deact Code Makes the next outbound call not secure It is redundant if all outbound calls are not secure by default The default is 19 Conference Act Code If this code is specified the user must enter it before dialing the third party for a conference call Enter the code for a conference call Attn Xfer Act Code If the code is specified...

Page 152: ... triggers the ATA device to perform a blind transfer to a target number that is prepended by the service code For example after the user dials 98 the ATA device plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number which is checked according to dial plan as in normal dialing When a complete number is entered the ATA device sends a blind REFER to the ...

Page 153: ... normal call This feature allows the proxy to process features like call forward 72 or BLock Caller ID 67 The codes should not conflict with any of the other vertical service codes internally processed by the ATA device You can empty the corresponding code that you do not want to ATA device to process You can add a parameter to each code in Features Dial Services Codes to indicate what tone to pla...

Page 154: ...vice announcements Service Annc Extension Codes Extension codes for service announcements Prefer G711u Code Makes this codec the preferred codec for the associated call The default is 017110 Force G711u Code Makes this codec the only codec that can be used for the associated call The default is 027110 Prefer G711a Code Makes this codec the preferred codec for the associated call The default is 017...

Page 155: ... Makes this codec the only codec that can be used for the associated call The default is 0272624 Prefer G726r32 Code Makes this codec the preferred codec for the associated call The default is 0172632 Force G726r32 Code Makes this codec the only codec that can be used for the associated call The default is 0272632 Prefer G726r40 Code Makes this codec the preferred codec for the associated call The...

Page 156: ...me hh stands for hours and mm stands for minutes Seconds are optional Time Zone Selects the number of hours to add to GMT to generate the local time for caller ID generation Choices are GMT 12 00 GMT 11 00 GMT GMT 01 00 GMT 02 00 GMT 13 00 The default is GMT 08 00 FXS Port Impedance Sets the electrical impedance of the FXS port Choices are 600 900 600 2 16uF 900 2 16uF 270 750 150nF 220 850 120nF ...

Page 157: ...esired instead of addition The save time value is in this format HH mm ss The month value equals any value in the range 1 12 January December The day value equals any value in the range 1 31 If day is 1 it means the weekday on or before the end of the month in other words the last occurrence of weekday in that month The weekday value equals any value in the range 1 7 Monday Sunday It can also equa...

Page 158: ...ecimal places The range is 6 000 to 12 000 The Call Progress Tones and DTMF playback level are not affected by the FXS Port Output Gain parameter The default is 3 DTMF Playback Level Local DTMF playback level in dBm up to one decimal place The default is 16 0 DTMF Playback Length Local DTMF playback duration in milliseconds The default is 1 Detect ABCD To enable local detection of DTMF ABCD select...

Page 159: ...CID CIDCW and VMWI FSK sent after DTAS but no polarity reversal and before first ring Waits for ACK from CPE after DTAS for CIDCW ETSI FSK With PR UK CID CIDCW and VMWI FSK is sent after polarity reversal and DTAS and before first ring Waits for ACK from CPE after DTAS for CIDCW Polarity reversal is applied only if equipment is on hook DTMF Denmark With PR CID only DTMF sent after polarity reversa...

Page 160: ...ttings for the Line FXO ports 1 to 4 With some variations depending on the model this page includes the following sections Line Enable section section on page161 Streaming Audio Server SAS section section on page 162 NAT Settings section section on page 163 Network Settings section section on page164 SIP Settings section section on page165 Call Feature Settings section section on page168 Proxy and...

Page 161: ...In a configuration profile the Line parameters must be appended with the appropriate numeral for example 1 or 2 to identify the line to which the setting applies The number of lines varies with the model of the ATA device For example the SPA2102 provides two Line tabs Line 1 and Line 2 while the SPA8000 provides eight tabs Line1 through Line 8 Voice tab Line page Line Enable section Line Enable To...

Page 162: ... used for outgoing calls Instead it auto answers incoming calls and streams audio RTP packets to the caller The default is no SAS DLG Refresh Intvl If this value is not zero it is the interval at which the streaming audio server sends out session refresh SIP re INVITE messages to determine whether the connection to the caller is still active If the caller does not respond to the refresh message th...

Page 163: ...ear in the c line and the port number and if specified in the m line of the SDP If this value is not specified or equal to 0 then c 0 0 0 0 and a sendonly will be used in the SDP to tell the SAS client to not to send any RTP to this SAS line If a non zero value is specified then a sendrecv and the SAS client will stream audio to the given address Special case If the value is IP then the SAS line s...

Page 164: ...efault is 0xb8 RTP CoS Value 0 7 CoS value for RTP data The default is 6 Network Jitter Level Determines how jitter buffer size is adjusted by the ATA device Jitter buffer size is adjusted dynamically The minimum jitter buffer size is 30 milliseconds or 10 milliseconds current RTP frame size whichever is larger for all jitter level settings However the starting jitter buffer size value is larger f...

Page 165: ...ber of the SIP message listening and transmission port The default is 5060 SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses 18x and use of PRACK requests select yes Otherwise select no The default is no EXT SIP Port The external SIP port number Auth Resync Reboot If this feature is enabled the ATA device authenticates the sender whe...

Page 166: ...ine only for all messages 1 line excl OPT Logs the start line only for all messages except OPTIONS requests responses 1 line excl NTFY Logs the start line only for all messages except NOTIFY requests responses 1 line excl REG Logs the start line only for all messages except REGISTER requests responses 1 line excl OPT NTFY REG Logs the start line only for all messages except OPTIONS NOTIFY and REGI...

Page 167: ...inate stale call legs upon completion of call transfers Multiple delay settings Referor Refer Target Referee and Refer To Target are configured on this screen For the Referor Bye Delay enter the appropriate period of time in seconds The default is 4 Refer Target Bye Delay For the Refer Target Bye Delay enter the appropriate period of time in seconds The default is 0 Referee Bye Delay For the Refer...

Page 168: ... on the SPA2102 only Default is no Blind Attn Xfer Enable Enables the ATA device to perform an attended transfer operation by ending the current call leg and performing a blind transfer of the other call leg If this feature is disabled the ATA device performs an attended transfer operation by referring the other call leg to the current call leg while maintaining both call legs To use this feature ...

Page 169: ... the outbound proxy within a dialog Ignored if the parameter Use Outbound Proxy is no or the Outbound Proxy parameter is empty The default is yes Register Enable periodic registration with the Proxy parameter This parameter is ignored if Proxy is not specified The default is yes Make Call Without Reg Allow making outbound calls without successful dynamic registration by the unit If No dial tone wi...

Page 170: ...s after it has failed over to a lower priority server This parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the PAP2T will not attempt to fall back after a fail over T...

Page 171: ...e number for this line Call Capacity Maximum number of calls allowed on this line interface Choices unlimited 1 2 3 25 Default is 16 Note that the the ATA device does not distinguish between incoming and outgoing calls when talking about call capacity NOTE unlimited 16 Cfwd No Ans Delay Delay in seconds before the call forwarding of no answer calls feature is triggered The default is 20 Mini Certi...

Page 172: ... the ATA device On the SPA8800 these settings are configured on the Phone pages only Call Waiting Serv Enable Call Waiting Service The default is yes Block CID Serv Enable Block Caller ID Service The default is yes Block ANC Serv Enable Block Anonymous Calls Service The default is yes Dist Ring Serv Enable Distinctive Ringing Service The default is yes Cfwd All Serv Enable Call Forward All Service...

Page 173: ... Way Call Serv Enable Three Way Calling Service Three Way Calling is required for Three Way Conference and Attended Transfer The default is yes Three Way Conf Serv Enable Three Way Conference Service Three Way Conference is required for Attended Transfer The default is yes Attn Transfer Serv Enable Attended Call Transfer Service Three Way Conference is required for Attended Transfer The default is...

Page 174: ...711a and G711u On the other hand two G 723 1 G 726 resources are available per device Therefore it is important to disable the use of G 729a in order to guarantee the support of two simultaneous G 723 G 726 codec Secure Call Serv Enable Secure Call Service The default is yes Referral Serv Enable Referral Service See the Referral Services Codes parameter for more details The default is yes Feature ...

Page 175: ... 40 G729a or G723 The default is Unspecified Use Pref Codec Only To use only the preferred codec for all calls select yes The call fails if the far end does not support this codec Otherwise select no The default is no Silence Supp Enable To enable silence suppression so that silent audio frames are not transmitted select yes Otherwise select no The default is no Silence Threshold Select the approp...

Page 176: ...select no The default is yes FAX CED Detect Enable To enable detection of the fax Caller Entered Digits CED tone select yes Otherwise select no The default is yes G726 32 Enable To enable the use of the G 726 codec at 32 kbps select yes Otherwise select no The default is yes FAX CNG Detect Enable To enable detection of the fax Calling Tone CNG select yes Otherwise select no The default is yes G726...

Page 177: ...od to transmit DTMF signals to the far end InBand AVT INFO Auto InBand INFO or AVT INFO InBand sends DTMF using the audio path AVT sends DTMF as eypents INFO uses the SIP INFO method Auto uses InBand or AVT based on the outcome of codec negotiation The default is Auto DTMF Tx Mode DTMF Detection Tx Mode is available for SIP information and AVT Options are Strict or Normal The default is Strict for...

Page 178: ...es Otherwise select no The default is yes Hook Flash Tx Method Select the method for signaling hook flash events None AVT or INFO None does not signal hook flash events AVT uses RFC2833 AVT event 16 INFO uses SIP INFO with the single line signal hf in the message body The MIME type for this message body is taken from the Hook Flash MIME Type setting The default is None FAX Disable ECAN If enabled ...

Page 179: ... a caller or callee then select caller or callee If you want the Gateway to detect the fax tone only if the Gateway is the caller then select caller only If you want the Gateway to detect the fax tone only if the Gateway is the callee then select callee only FAX Tone Detect Mode This parameter has three possible values caller or callee SPA will detect FAX tone whether it is callee or caller caller...

Page 180: ...ch that when the user dials 1408 7digits the call will be routed to Gateway 1 Without the gw1 syntax all calls are routed to the given proxy by default except IP dialing The default is blank GW1 2 3 4 NAT Mapping Enable If enabled the ATA device uses NAT mapping when contacting Gateway 1 The default is no GW1 2 3 4 Auth ID This value is the authentication user id to be used by the SPA to authentic...

Page 181: ...ght dial plans The dial plans in this pool can be associated with a VoIP Caller or a PSTN Caller The dial plan syntax is consistent for all fields The default dial plan script for each line is as follows xx 3469 11 0 00 2 9 xxxxxx 1xxx 2 9 xxxxxx xxxxxxxxxxxx The syntax for a dial plan expression is as follows Dial Plan Entry Functionality xx Allow arbitrary 2 digit star code 3469 11 Allow x11 seq...

Page 182: ... where gw0 represents the local PSTN gateway in the same SPA3102 Example 1 1xxxxxxxxxx fwdnat pulver com 5082 uid jsmith pwd xy z Example 2 1xxxxxxxxxx fwd pulver com nat uid jsmith pwd xyz Example 3 39 11 gw0 Enable IP Dialing Enable or disable IP dialing If IP dialing is enabled one can dial user id a b c d port where and are dialed by entering user id must be numeric like a phone number and a b...

Page 183: ... no emergency number Maximum number length is 63 characters The default is blank Idle Polarity Polarity before a call is connected Forward or Reverse The default is Forward Caller Conn Polarity Polarity after an outbound call is connected Forward or Reverse The default is Forward Callee Conn Polarity Polarity after an inbound call is connected Forward or Reverse The default is Forward Line 1 VoIP ...

Page 184: ...ne 1 2 3 4 5 6 7 8 The default is 1 PSTN CID for VoIP CID Choose yes or no The default is no PSTN CID Number Prefix The prefix to add to the caller ID number on the PBX to ensure that a callback goes to the correct number Enter the digits PSTN caller Default DP Index of the dial plan in the dial plan pool to be used when the PSTN Caller is not authenticated Choose from 1 2 3 4 5 6 7 8 The default ...

Page 185: ...till up If value is set to 0 SPA will not send refresh messages and VoIP call leg status is not checked by the SPA The refresh message is a SIP ReINVITE and the VoIP peer must response with a 2xx response If VoIP peer does not reply or response is not greater than 2xx the SPA will disconnect both PSTN and VoIP call legs automatically The range is 0 255 The default is 30 PSTN Ring Timeout Delay aft...

Page 186: ...y transmits digits through the Line FXO port The syntax is on time off time expressed in seconds with up to two decimal places The permitted range is 0 05 to 3 00 The default is 1 1 PSTN Hook Flash Len Default is 25 Detect CPC CPC is a brief removal of tip and ring voltage If enabled the SPA will disconnect both call legs when this signal is detected during a gateway call The default is yes Detect...

Page 187: ...ty in seconds to trigger a gateway call disconnection if Detect VoIP Long Silence is yes The default is 30 PSTN Silence Threshold This parameter adjusts the sensitivity of PSTN silence detection Choose from very low low medium high very high The higher the setting the easier to detect silence and hence easier to trigger a disconnection The default is medium Min CPC Duration Specify the minimum dur...

Page 188: ...ion of the segment set is interpreted as the minimum duration of the tone to trigger detection 6 segments of on off time seconds can be specified A 10 margin is used to validated cadence characteristics of the tone Disconnect Tone continued The Disconnect Tone Script values for various countries follow US 480 30 620 30 4 25 25 1 2 UK 400 30 400 30 2 3 0 1 2 France 440 30 440 30 2 0 5 0 5 1 2 Germa...

Page 189: ...750 150nF Australia 220 820 120nF New Zealand 370 620 310nF Ring Frequency Min The lower limit of the ring frequency used to detect the ring signal The default is 10 SPA To PSTN Gain dB of digital gain or attenuation if negative to be applied to the signal sent from the SPA to the PSTN side The range is 15 to 12 The default is 0 Ring Frequency Max The higher limit of the ring frequency used to det...

Page 190: ... are 10 12 14 16 The default is 10 mA Ring Timeout Choose from 0 128 256 384 512 640 768 896 1024 1152 1280 1408 1536 1664 1792 1920 ms The default is 640 ms On Hook Speed Choose from Less than 0 5ms 3ms ETSI 26ms Australia The default is Less than 0 5ms Ring Threshold Choose from 13 5 16 5 19 35 2 65 40 5 49 5 Vrms The default is 13 5 16 5 Vrms Current Limiting Enable Enable or disable current li...

Page 191: ...ction on page191 SIP Settings section section on page192 Subscriber Information section section on page195 Dial Plan section section on page197 Proxy and Registration section section on page 205 Voice tab Trunk Group page SPA8000 Line Enable section Voice tab Trunk Group page SPA8000 Network Settings section Line Enable To enable this line for service select yes Otherwise select no The default is ...

Page 192: ...Port number of the SIP message listening and transmission port The default is 5060 SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses 18x and use of PRACK requests select yes Otherwise select no The default is no Auth Resync Reboot If this feature is enabled the ATA device authenticates the sender when it receives the NOTIFY resync re...

Page 193: ...gging 1 line Logs the start line only for all messages 1 line excl OPT Logs the start line only for all messages except OPTIONS requests responses 1 line excl NTFY Logs the start line only for all messages except NOTIFY requests responses 1 line excl REG Logs the start line only for all messages except REGISTER requests responses 1 line excl OPT NTFY REG Logs the start line only for all messages e...

Page 194: ...und Proxy if Use Outbound Proxy is yes The default is no Referor Bye Delay Controls when the ATA device sends BYE to terminate stale call legs upon completion of call transfers Multiple delay settings Referor Refer Target Referee and Refer To Target are configured on this screen For the Referor Bye Delay enter the appropriate period of time in seconds The default is 4 Refer Target Bye Delay For th...

Page 195: ...no Auth ID Authentication ID for SIP authentication Call Capacity Maximum number of calls allowed on this trunk group Choices 1 15 or unlimited 16 calls Default is unlimited Both incoming call and outgoing call are counted towards this limit The call capacity has the following impact on call handling Inbound calls When the limit is reached the Trunk SUA replies 486 to the caller Outbound calls Whe...

Page 196: ...the moment 1 hunt al 30 0 cfwd 14089993326 A wildcard character is used to represent all trunk lines All lines ring simultaneously hunt al If there is no answer after 30 seconds 30 the call is forwarded to the specified number cfwd 14089993326 hunt ra 12 1 cfwd 14089993326 A wildcard character is used to represent all trunk lines The Trunk SUA hunts in random order hunt ra If there is no answer wi...

Page 197: ...umber 0 9A D 0 9A D NAT Mapping Enable To use externally mapped IP addresses and SIP RTP ports in SIP messages select yes Otherwise select no The default is no NAT Keep Alive Enable To send the configured NAT keep alive message periodically select yes Otherwise select no The default is no NAT Keep Alive Msg Enter the keep alive message that should be sent periodically to maintain the current NAT m...

Page 198: ...roxy parameter This parameter is ignored if Proxy is not specified The default is yes Make Call Without Reg Allow making outbound calls without successful dynamic registration by the unit If No dial tone will not play unless registration is successful The default is no Register Expires Allow answering inbound calls without successful dynamic registration by the unit If proxy responded to REGISTER ...

Page 199: ...rvers after it has failed over to a lower priority server This parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the PAP2T will not attempt to fall back after a fail ov...

Page 200: ...TTP Authentication section section on page 213 FXO PSTN Timer Values sec section section on page 214 PSTN Disconnect Detection section section on page 216 International Control Settings section section on page 219 Voice tab PSTN Line page SPA3102 Line Enable section Voice tab PSTN Line page SPA3102 NAT Settings section Line Enable To enable this line for service select yes Otherwise select no The ...

Page 201: ... is REGISTER a REGISTER message without contact is sent Escape sequence of xx is also accepted For example 0d 0a is unescaped into r n CRLF The default is NOTIFY NAT Keep Alive Dest Destination that should receive NAT keep alive messages If the value is PROXY the messages are sent to the current or outbound proxy The default is PROXY SIP ToS DiffServ Value TOS DiffServ field value in UDP IP packet...

Page 202: ...r buffer should be adjusted Select the appropriate setting up and down up only down only or disable The default is up and down SIP Port Port number of the SIP message listening and transmission port The default is 5060 SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses 18x and use of PRACK requests select yes Otherwise select no The d...

Page 203: ...sages to log Choices are as follows none No logging 1 line Logs the start line only for all messages 1 line excl OPT Logs the start line only for all messages except OPTIONS requests responses 1 line excl NTFY Logs the start line only for all messages except NOTIFY requests responses 1 line excl REG Logs the start line only for all messages except REGISTER requests responses 1 line excl OPT NTFY R...

Page 204: ...ror Bye Delay Controls when the ATA device sends BYE to terminate stale call legs upon completion of call transfers Multiple delay settings Referor Refer Target Referee and Refer To Target are configured on this screen For the Referor Bye Delay enter the appropriate period of time in seconds The default is 4 Refer Target Bye Delay For the Refer Target Bye Delay enter the appropriate period of time...

Page 205: ... This parameter is ignored if the Proxy parameter is not specified The default is yes Make Call Without Reg Allow making outbound calls without successful dynamic registration by the unit If No dial tone will not play unless registration is successful The default is no Register Expires Allow answering inbound calls without successful dynamic registration by the unit If proxy responded to REGISTER ...

Page 206: ... parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the PAP2T will not attempt to fall back after a fail over The default is 3600 Proxy Redundancy Method The PAP2T makes...

Page 207: ...ce Therefore it is important to disable the use of G 729a in order to guarantee the support of two simultaneous G 723 G 726 codec Use Auth ID To use the authentication ID and password for SIP authentication select yes Otherwise select no to use the user ID and password The default is no Auth ID Authentication ID for SIP authentication Call Capacity Maximum number of calls allowed on this line inte...

Page 208: ... G723 Enable To enable the use of the G723a codec at 6 3 kbps select yes Otherwise select no The default is yes Echo Canc Adapt Enable To enable the echo canceller to adapt select yes Otherwise select no The default is yes G726 16 Enable To enable the use of the G726 codec at 16 kbps select yes Otherwise select no The default is yes Echo Supp Enable To enable the use of the echo suppressor select ...

Page 209: ...odec Symmetric To force the ATA device to use a symmetric codec during fax passthrough select yes Otherwise select no The default is yes DTMF Process AVT This field is not available for the PAP2T To use the DTMF process AVT feature select yes Otherwise select no The default is yes FAX Passthru Method Select the fax passthrough method None NSE or ReINVITE The default is NSE DTMF Tx Method Select th...

Page 210: ...odec negotiation on the first call so that other codecs can be used for the second line To use this feature select yes Otherwise select no The default is yes FAX Enable T38 To enable the use of the ITU T T 38 standard for faxing select yes Otherwise select no The default is yes FAX Tone Detect Mode This parameter has three possible values caller or callee SPA will detect FAX tone whether it is cal...

Page 211: ... Plan 1 to 8 The dial plan syntax is the same as that used for Line 1 See Dial Plan section on page 181 VoIP To PSTN Gateway Enable Enable or disable VoIP To PSTN Gateway functionality The default is yes VoIP Caller Authentication Method Method to be used to authenticate a VoIP Caller to access the PSTN gateway Choose from none PIN HTTP Digest The default is none VoIP PIN Max Retry Number of trial...

Page 212: ...is down Choose from none 1 2 3 4 5 6 7 8 The default is 1 VoIP Caller ID Pattern A comma separated list of caller number templates such that callers with numbers not matching any of these templates are rejected for PSTN gateway service regardless of the setting of the authentication method The comparison is applied before the access list is applied If this parameter is blank not specified all call...

Page 213: ...he credentials are missing or incorrect the SPA will challenge the caller with a 401 response The VoIP caller whose authentication user id equals to this ID is referred to VoIP User 1 of this SPA NOTE If the caller specifies an authentication user id that does not match any of the VoIP User Auth ID s the INVITE will be rejected with a 403 response The default is blank VoIP User 1 2 3 4 5 6 7 8 DP ...

Page 214: ...git Timeout Timeout to wait for the 1st or subsequent PIN digits from a VoIP caller The range is 0 255 The default is 10 PSTN PIN Digit Timeout Timeout to wait for the 1st or subsequent PIN digits from a PSTN caller The range is 0 255 The default is 10 VoIP DLG Refresh Intvl Interval between SIP Dialog refresh messages sent by the SPA to detect if the VoIP call leg is still up If value is set to 0...

Page 215: ...a PSTN number The range is 0 255 The default is 1 PSTN Ring Timeout Delay after a ring burst before the SPA decides that PSTN ring has ceased The range is 0 255 The default is 5 PSTN Dial Digit Len Determines the on off time when transmitting digits through the FXO port The syntax is on time off time where on time and off time are expressed in seconds with up to two decimal places within the permi...

Page 216: ... 3 00 The default is 1 1 PSTN Hook Flash Len Default is 25 Detect CPC CPC is a brief removal of tip and ring voltage If enabled the SPA will disconnect both call legs when this signal is detected during a gateway call The default is yes Detect Polarity Reversal If enabled SPA will disconnect both call legs when this signal is detected during a gateway call If it is a PSTN gateway call the 1st pola...

Page 217: ...meter which depends on the region of the PSTN service The default is yes PSTN Long Silence Duration This value is minimum length of PSTN silence or inactivity in seconds to trigger a gateway call disconnection if Detect Long Silence is yes The default is 30 Silence Threshold This parameter adjusts the sensitivity of PSTN silence detection Choose from very low low medium high very high The higher t...

Page 218: ...lowed Total duration of the segment set is interpreted as the minimum duration of the tone to trigger detection 6 segments of on off time seconds can be specified A 10 margin is used to validated cadence characteristics of the tone The Disconnect Tone Script values for various countries follow US 480 30 620 30 4 25 25 1 2 UK 400 30 400 30 2 3 0 1 2 France 440 30 440 30 2 0 5 0 5 1 2 Germany 440 30...

Page 219: ...nF New Zealand 370 620 310nF Ring Frequency Min Specify the lower limit of the ring frequency used to detect the ring signal The default is 10 SPA To PSTN Gain dB of digital gain or attenuation if negative to be applied to the signal sent from the SPA to the PSTN side The range is 15 to 12 The default is 0 Ring Frequency Max Specify the higher limit of the ring frequency used to detect the ring si...

Page 220: ...100 Ring Validation Time Choose from 100 150 200 256 384 512 640 1024 ms The default is 256ms Ring Indication Delay Choose from 0 512 768 1024 1280 1536 1792 ms The default is 512ms Ring Timeout Choose from 0 128 256 384 512 640 768 896 1024 1152 1280 1408 1536 1664 1792 1920 ms The default is 640 ms Ring Threshold Choose from 13 5 16 5 19 35 2 65 40 5 49 5 Vrms The default is 13 5 16 5 Vrms Ringe...

Page 221: ... the following sections Call Forward Settings section section on page 222 Selective Call Forward Settings section section on page 223 Speed Dial Settings section section on page 223 Supplementary Service Settings section section on page 224 Distinctive Ring Settings section section on page 225 Ring Settings section section on page 226 NOTE When a call is made from Line 1 or Line 2 the ATA device s...

Page 222: ...calA through the PSTN gateway The default is blank Cfwd Busy Dest Forward number for Call Forward Busy Service Same as Cfwd All Dest The default is blank Cfwd No Ans Dest Forward number for Call Forward No Answer Service Same as Cfwd All Dest In addition to normal call forward destination as used in the other ATAs on the SPA3102 you can specify the following additional parameters gw0 forward the c...

Page 223: ...lective 1 2 3 4 5 6 7 or 8 Same as Cfwd All Dest The default is blank Block Last Caller ID of caller blocked via the Block Last Caller service The default is blank Accept Last Caller ID of caller accepted via the Accept Last Caller service The default is blank Cfwd Last Caller The Caller number that is actively forwarded to Cfwd Last Dest by using the Call Forward Last activation code The default ...

Page 224: ...lt is yes Block CID Setting Block Caller ID on off for all calls The default is no Block ANC Setting Block Anonymous Calls on or off The default is no DND Setting DND on or off The default is no CID Setting Caller ID Generation on or off The default is yes CWCID Setting Call Waiting Caller ID Generation on or off The default is yes Dist Ring Setting Distinctive Ring on or off The default is yes Se...

Page 225: ...t and the call must be picked up manually before loopback starts The default is Automatic Media Loopback Mode The loopback mode to assume locally when making call to request media loopback Choices are Source and Mirror Default is Source Note that if the ATA device answers the call the mode is determined by the caller Media Loopback Type The loopback type to use when making call to request media lo...

Page 226: ...ded 0 10 0s The default is 0 Cblk Ring Splash Len Duration of ring splash when a call is blocked 0 10 0s The default is 0 VMWI Ring Splash Len Duration of ring splash when new messages arrive before the VMWI signal is applied 0 10 0s The default is 5 VMWI Ring Policy The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the ATA device indicating the...

Page 227: ...ce tab PSTN User page SPA3102 PSTN To VoIP Selective Call Forward Settings section Ring On No New VM If enabled the ATA device will play a ring splash when the VM server sends SIP NOTIFY message to the ATA device indicating that there are no more unread voice mails Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp The default is no Cfwd Sel1 8 Caller Eight PSTN C...

Page 228: ...Ring Thru Line 1 Ring Settings section Speed Dial 1 9 The VoIP number to call when the PSTN caller dials a single digit 2 Ring1 8 Caller Eight PSTN Caller Number Patterns such that the corresponding ring will be used to ring through Line 1 if the PSTN caller matches this pattern Default Ring The default ring to be used to ring through Line 1 Choose from 1 2 3 4 5 6 7 8 Follow Line 1 If Follow Line...

Page 229: ...e administration web server The address I entered did not work A Use the Interactive Voice Response Menu to find out the Internet IP address Follow these steps 1 Use a telephone connected to the Phone 1 port of the ATA device 2 Press in other words press the star key four times 3 After the greeting plays press 110 4 Write down the IP address as it is announced 5 Press 7932 6 Press 1 to enable WAN ...

Page 230: ...ire HTML page Alternatively from your browser you can select File Save as HTML from any of the administration web server pages Do this in Admin Advanced mode This saves all the tabs into one HTML file This HTML file is helpful to provide to our support team when you have a problem or technical question Q How do I debug my ATA device Is there a syslog A The ATA devices send out debug information vi...

Page 231: ...t to the factory default settings The Admin account password will be reset to the default of blank 2 Press 1 to confirm the operation Press to cancel the operation 3 Log in to the unit using the User or Admin account without a password and reconfigure the unit as necessary Q My ATA device is behind a NAT device or firewall and I m unable to make a call or I m only receiving a one way connection Wh...

Page 232: ...nt behind a NAT device to find out its public address the type of NAT it is behind and the port associated on the Internet connection with a particular local port This information is used to set up UDP communication between two hosts that are both behind NAT routers Open source STUN software can be obtained at the following website http www voip info org wiki Open Source VOIP Software NOTE STUN do...

Page 233: ...235 SPA8800 on page 235 WRTP54G on page 236 PAP2T Device Dimensions 3 98 x 3 98 x 1 10 101 x 101 x 28 mm W x H x D Unit Weight 5 40 oz 153g Power 100 240V 50 60Hz AC Input Certification FCC Part 15 Class B cUL CE IC 003 A Tick Operating Temp 32 to 113º F 0 to 45ºC Storage Temp 17º to 158ºF 27 to 70ºC Operating Humidity 10 to 90 relative humidity Non Condensing Storage Humidity 10 to 90 relative hu...

Page 234: ...p 13º to 185ºF 25 to 85ºC Operating Humidity 10 to 90 relative humidity Non Condensing Storage Humidity 10 to 90 relative humidity Non Condensing Device Dimensions 3 98 x 3 98 x 1 10 101 x 101 x 28 mm Unit Weight 5 11 oz 0 145kg Power 100 240V 50 60Hz 26 34VA AC Input Certification FCC Part 15 Class B CE ICES 003 A Tick Certification RoH Operating Temp 32º to 113º F 0 to 45ºC Storage Temp 13º to 1...

Page 235: ...rage Temp 13º to 185ºF 25 to 85ºC Operating Humidity 10 to 90 relative humidity Non Condensing Storage Humidity 10 to 90 relative humidity Non Condensing Device Dimensions 6 69 x 1 54 x 8 66 170 x 39 x 220 mm Unit Weight 2 85 lbs 1 30kg Power 100 240V 50 60Hz 26 34VA AC Input Certification FCC Part 15 Class B CE ICES 003 A Tick Certification RoH UL Operating Temp 32º to 113º F 0 to 45ºC Storage Te...

Page 236: ... x 6 69 x 1 22 170 x 170 x 31 mm Unit Weight 13 60 oz 39 kg Power External 12V DC 1 0A Certification FCC Part 15 Class B CE UL Operating Temp 32º to 104º F 0 to 40ºC Storage Temp 4º to 140ºF 20 to 60ºC Operating Humidity 10 to 85 relative humidity Non Condensing Storage Humidity 5 to 90 relative humidity Non Condensing ...

Page 237: ...ot require it If you use an older web browser you may have to add http in front of the web address Resource Location Technical Documentation http www cisco com en US products ps10024 prod_maintenance_guides_list html Firmware Downloads Go to tools cisco com support downloads and enter the model number in the Software Search box Cisco Community Central Small Business Support Community www myciscoco...

Page 238: ...om en US products ps10024 prod_maintenance_guides_list html Cisco Partner Central Login Required www cisco com web partners sell smb Cisco Small Business Home www cisco com smb Resource Location Document Title Description Intended Audience SPA9000 Voice System Installation and Configuration Guide Using the Setup Wizard Installation configuration and maintenance of the SPA9000 Voice System by using...

Page 239: ...stration Guide Configuration and management of SPA9x2 series IP phones Deployment options with or without the SPA9000 IP PBX SPA9x2 series IP phones VARs and Service Providers SPA9x2 Phone User Guide Phone setup Phone features SPA9x2 series IP phones VARs and phone end users Analog Telephone Adapter Administration Guide Administration and use of Cisco Small Business ATAs PAP2T SPA2102 SPA3102 SPA8...

Page 240: ...ete the following steps a Select all files in the book b Choose File Import Formats c Import From Document Choose this file ATA_Variables fm d Click Deselect All to clear all of the check boxes and then check only the following check boxes Variable Defintions Conditional Text Settings e Click Import When you are ready to publish STEP 1 On the menu choose Special Conditional Text settings Click the...

Page 241: ...eck only the following check boxes Variable Defintions Conditional Text Settings e Click Import STEP 3 Generate update your book to ensure that all page numbers chapter numbers etc are updated With shared files the files will pick up the appropriate document numbering from the book file during generate update process ...

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