SIP 2.0 Administrator’s Guide SoundPoint®/SoundStation® IP SIP Version 2.0.3B Addendum Version 2.1 Addendum January 2007 Copyright © 2007 Polycom, Inc. All rights reserved.
Notices 1. Specifications subject to change without notice. Polycom, Inc. 1565 Barber Lane, Milpitas CA 95035, USA www.polycom.com Part Number: 1725-11530-210 Rev A Copyright © 2007 Polycom, Inc. All rights reserved.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 1Addendum This addendum addresses changes to the SoundPoint IP / SoundStation IP SIP 2.0 Administrator’s Guide made by the release of the SoundPoint IP 650 phone. The SoundPoint IP 650 phone behaves in a similar manner to the SoundPoint IP 601 (supports the SoundPoint IP Expansion Module) unless otherwise specified. For more information, refer to the Release Notes for the SIP Application, Version 2.0.3 B. Note The various .hd.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum Attribute Default voice.headset.rxag.adjust.IP_650 1 voice.headset.txag.adjust.IP_650 18 voice.headset.sidetone.adjust.IP_650 -3 Important Polycom recommends that you do not change these values. 1.1.3 LCD Backlight Backlight intensity on the SoundPoint IP 650 phone has three modes: • Backlight On • Backlight Idle • Dim You can modify the Backlight On intensity and the Backlight Idle intensity separately.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 1.1.4 Expanded Memory and Expanded Flash Memory Changes can be found in the following parameters in the sip.cfg configuration file: • Directory
Attribute Permitted Values Default Interpretation dir.local.volatile.maxSize 1 to 100 100 Maximum size in Kbytes of volatile storage that the directory will be permitted to consume. dir.local.volatile.8meg 0, 1 0 Attribute applies only to platforms with 8 Mbytes of flash memory.Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum • RAM Disk Permitted Values Attribute ramdisk.bytesPerBlock 0, 32, 33, ..., 1024 Default Interpretation 0 These three parameters use internal defaults when value is set to 0. • Finder Permitted Values Attribute 4 Default Interpretation res.finder.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 1.1.5 MicroBrowser The SoundPoint IP 650 phones support an XHTML microbrowser. This can be launched by pressing the Services key. MicroBrowser parameter changes in the sip.cfg configuration file are as follows: Attribute Permitted Values Default Interpretation mb.limits.nodes positive integer 256 Limits the number of tags which the XML parser will handle. This limits the amount of memory used by complicated pages.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 1.1.6 G.722 Audio Codec The SoundPoint IP 650 supports the G.722 audio codec. Changes can be found in the following parameters in the sip.cfg configuration file: • Codec Preferences Attribute voice.codecPref.IP_650.G711M u Permitted Values Default Interpretation Null, 1-3 2 Specifies the codec preferences for the SoundPoint IP 650 platform. 1 = highest 3 = lowest Null = do not use voice.codecPref.IP_650.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Permitted Values Attribute Addendum Default Interpretation voice.audioProfile.G722.jitterBufferShrink 10, 20, 30, ... (multiple of 10) 500 The absolute minimum duration time (in milliseconds) of RTP packet Rx with no packet loss between jitter buffer size shrinks. Use smaller values (1000 ms) to minimize the delay on known good networks. Use larger values to minimize packet loss on networks with large jitter (3000 ms). voice.audioProfile.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum • Receive Attribute Default voice.rxEq.hs.IP_650.preFilter.enable 1 voice.rxEq.hs.IP_650.postFilter.enable 0 voice.rxEq.hd.IP_650.preFilter.enable 1 voice.rxEq.hd.IP_650.postFilter.enable 0 voice.rxEq.hf.IP_650.preFilter.enable 1 voice.rxEq.hf.IP_650.postFilter.enable 0 Important Polycom recommends that you do not change these values. • Transmit . Attribute Default voice.txEq.hs.IP_650.preFilter.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 1.1.7 USB Diagnostics The SoundPoint IP 650 phone has a USB port, which will be supported by future releases of the SIP application. USB port parameters can be found in the USB parameter in the sip.cfg configuration file. Attribute Permitted Values Default Interpretation usb.enable 0, 1 0 This parameter enables or disables the USB port on the phone. usb.bulkDrive.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP 10 Copyright © 2006 Polycom, Inc.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 2 Addendum This addendum addresses changes to the SoundPoint IP / SoundStation IP SIP 2.0 Administrator’s Guide made by the release of the SIP 2.1 application. For more information, refer to the Release Notes for the SIP Application, Version 2.1 . Note The various .hd. parameters in sip.cfg (such as voice.aec.hd.enable, voice.ns.hd.enable, and voice.agc.hd.enable) are headset parameters.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 2.1.4 Server Redundancy Server redundancy enhancements provides backup to other SIP server(s) by providing basic registration and redirection services. Refer to the “Technical Bulletin 5844: SIP Server Fallback Enhancements on SoundPoint® IP Phones” at www.polycom.com/ support/voip/ . 2.1.5 MicroBrowser An XHTML microBrowser is now supported on the SoundPoint IP 430 and 501 phones.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 2.1.6 Disable Message Waiting Indicator by Registration The SIP application has been enhanced to allow the message waiting indicator to be disabled by registration. • Changes can be found in the following parameters in the phone1.cfg configuration file: Attribute msg.mwi.x.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Attribute tcpIpApp.sntp.daylightSavings.fixedDayEnable Addendum Permitted Values Default Interpretation 0, 1 0 If set to 0, month, date, and dayOfWeek are used in DST start date calculation. If set to 1, then only month and date are used. tcpIpApp.sntp.daylightSavings.start.month 1-12 3 (March) Month to start DST. Mapping: 1=Jan, 2=Feb, ..., 12=Dec tcpIpApp.sntp.daylightSavings.start.date 1-31 8 Day of the month to start DST.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum Attribute Permitted Values Default Interpretation tcpIpApp.sntp.daylightSavings.stop.date 1-31 1 Day of the month to stop DST. tcpIpApp.sntp.daylightSavings.stop.time 0-23 2 Time of day to stop DST in 24 hour clock. tcpIpApp.sntp.daylightSavings.stop.dayOfWeek 1-7 1 Day of week to stop DST. tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 2.1.9 Miscellaneous Configuration File Changes 2.1.9.1 sip.cfg The following changes have also occurred in the sip.cfg configuration file: Attribute Permitted Values Default Interpretation voIpProt.SIP.useSendonlyHold 0, 1 1 If set to 1, the phone will send a reinvite with a stream mode attribute of “sendonly” when a call is put on hold. This is the same as the previous behavior.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Attribute voIpProt.server.x.transport Addendum Permitted Values DNSnaptr or TCPpreferred or UDPOnly or TLS or TCPOnly Copyright © 2006 Polycom, Inc. Default Interpretation DNSnapt r If set to Null or DNSnaptr: If voIpProt.server.x.address is a hostname and voIpProt.server.x.port is 0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If voIpProt.server.x.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Permitted Values Attribute Default Interpretation DNSnapt r If set to Null or DNSnaptr: If voIpProt.SIP.outboundProxy.address is a hostname and voIpProt.SIP.outboundProxy.port is 0 or Null, do NAPTR then SRV lookups to try to discover the transport, ports and servers, as per RFC 3263. If voIpProt.SIP.outboundProxy.address is an IP address, or a port is given, then UDP is used.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum Attribute Permitted Values Default Interpretation call.stickyAutoLineSeize.onHookDialing Null, 0, 1 Null If call.stickyAutoLineSeize is set to 1, this parameter has no effect. The regular stickyAutoLineSeize behavior is followed. If call.stickyAutoLineSeize is set to 0 or Null and this parameter is set to 1, this overrides the stickyAutoLineSeize behavior for hot dial only. (Any New Call scenario seizes the next available line.
Administrator’s Guide - SoundPoint® IP / SoundStation® IP Addendum 2.1.9.3 device Parameter The following changes has also occurred in the device parameter: 10 Attribute Permitted Values Default Interpretation device.prov.redunAttemptLimit 10, Null 10 Refer to the File Transmit Tries parameter in 2.2.1.3.3 Server Menu on page 11 of the SIP 2.0 Administrator’s Guide. device.prov.redunInterAttemptDelay 300, Null 300 Refer to the Retry Wait parameter in 2.2.1.3.
Technical Bulletin 11572 Changes to Local Digit Maps on SoundPoint® IP Phones This technical bulletin provides detailed information on how to modify the configuration files to automate the setup phase of number-only calls. This information applies to SoundPoint IP phones running SIP application version 2.1 or later. Introduction Enhancements have been made to this feature that can eliminate the need for using the Dial or Send soft key when making outgoing calls.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Local Web Server Specify impossible match behavior, trailing # behavior, digit map matching strings, and time out value. (if enabled) Navigate to: http:///appConf.htm#ls Changes are saved to local flash and backed up to -phone.cfg on the boot server. Changes will permanently override global settings unless deleted through the Reset Local Config menu selection.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Attribute Permitted Values Default Interpretation dialplan.applyToUserSend 0, 1 1 This attribute covers the case when the user presses the Send soft key to send the dialed number. Value interpretation is the same as for dialplan.applyToCallListDial. dialplan.impossibleMatchHandlin g 0, 1 or 2 0 If set to 0, the digits entered up to and including the point where an impossible match occurred are sent to the server immediately.
Technical Bulletin SoundPoint ® IP, SIP 2.1 The following guidelines should be noted: • You must use only *, #, or 0-9 between second and third R • If a digit map does not comply, it is not included in the digit plan as a valid one. That is, no matching is done against it. • There is no limitation on the number of R triplet sets in a digit map. However, a digitmap that contains less than full number of triplet sets (for example, a total of 2Rs or 5Rs) is considered an invalid digit map.
Technical Bulletin SoundPoint ® IP, SIP 2.1 In the following tables, x is the registration number. IP 300, 301, and 430: x=1-2; IP 500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1 Attribute Permitted Values Default Interpretation dialplan.x.applyToCallListDial 0, 1 0 When present, and if dialplan.x.digitmap is not Null, this attribute overrides the global dial plan defined in the sip.cfg configuration file.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Attribute Permitted Values Default Interpretation dialplan.x.impossibleMatchHandling 0, 1 or 2 0 When present, and if dialplan.x.digitmap is not Null, this attribute overrides the global dial plan defined in the sip.cfg configuration file. For interpretation, refer to Dial Plan in Application Configuration File on page 2. dialplan.x.removeEndOfDial 0, 1 1 When present, and if dialplan.x.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Attribute dialplan.x.digitmap.timeOut Permitted Values string of positive integers separated by ‘|’ Default Interpretation Null When present, and if dialplan.x.digitmap is not Null, this attribute overrides the global dial plan defined in the sip.cfg configuration file. For more information, refer to Digit Map on page 3. Trademark Information Polycom®, SoundPoint®, and the Polycom logo design are registered trademarks of Polycom, Inc. in the U.
Technical Bulletin 9268 Billing Code Entry on SoundPoint® IP Phones with Sylantro This technical bulletin provides detailed information on how the SIP application has been modified for billing code entry when managed by a Sylantro call server. This information applies to SoundPoint IP phones running SIP application version 2.1 or later. Introduction Note This feature is only supported on Sylantro call servers.
Technical Bulletin SoundPoint ® IP, SIP 2.1 You may need to press the Send soft key to indicate you are finished entering the number. The cursor pauses after the last digit has been entered. The call is not placed at this time. A secondary dial tone is played and the text “Enter more digits” appears on the display just above the soft keys. 2. Enter the billing code. If the billing code is accepted, the call is placed at this time.
Technical Bulletin 17124 Syslog on SoundPoint® IP Phones This technical bulletin provides detailed information on how the SIP application has been modified to support logging system level messages and error conditions with communications networks to a centralized location. This information applies to SoundPoint IP phones running SIP application version 2.1 or later. Introduction Syslog is a de facto standard for forwarding log messages in an IP network.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Network Configuration Changes The Network Configuration menu on the SoundPoint IP phone running SIP 2.1 has been modified to include: • Syslog Menu To access the Syslog menu: 1. From the idle display on a SoundPoint IP phone, press the Menu key. 2. Using the Down Arrow key and the Select soft key, select Settings > Advanced > Admin Settings > Network Configuration. You must enter the administrative password to access this menu. The default value is “456”. 3.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Flash Parameter Configuration The global device parameter has been modified to include the following: Name Possible Values Description device.syslog.serverName dotted-decimal IP address OR domain name string The syslog server IP address or host name. None=0, UDP=1, TCP=2 The protocol that the phone will use to write to the syslog server. 0 to 23 A description of what generated the log message. For more information, refer to section 4.1.1 of RFC 3165.
Technical Bulletin 5844 SIP Server Fallback Enhancements on SoundPoint® IP Phones This technical bulletin provides detailed information on how the SIP application has been enhanced to support SIP server fallback. This information applies to SoundPoint IP phones running SIP application version 2.1 or later.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Terminology Before you read this document, it is important to understand certain terminology and become familiar with the server/registration configuration as described in the references listed in the References on page 8. The behavior described in this document supersedes that described in section 3.6.5 of the SIP 2.0 Administrator's Guide. SIP Registrations: SoundPoint IP phones support the ability to have multiple SIP Registrations per phone.
Technical Bulletin SoundPoint ® IP, SIP 2.1 SIP 2.1 Server Fallback Implementation In the SIP 2.1 release, the redundancy behavior of Polycom SoundPoint IP and SoundStation IP phones has been changed and improved by adding the ability for a single SIP Registration (Line) to be registered to more than one server concurrently. In previous releases, the phone would only maintain one active server registration per SIP Registration (Line).
Technical Bulletin SoundPoint ® IP, SIP 2.1 Phone Configuration The phones at the customer site are configured as follows: • Server 1 (the primary server) will be configured with the address of the service provider call server. The IP address of the server(s) to be used will be provided by the DNS server. For example: reg.1.server.1.address="voipserver.serviceprovider.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Behavior When the Primary Server Connection Fails For Outgoing Calls (INVITE Fallback) When the user initiates a call, the phone will go through the following steps to connect the call: 1. Try to make the call using the working server. 2. If the working server does not respond correctly to the INVITE, then try and make a call using the next server in the list (even if there is no current registration with these servers).
Technical Bulletin SoundPoint ® IP, SIP 2.1 Changes From Previous Phone Behavior (Releases Before SIP 2.1) Before SIP 2.1 In SIP 2.1 A Line is only capable of maintaining one server registration. A Line will maintain registrations with all servers that are configured as registrar servers. If two servers are configured (for example, reg.1.server.1.address = "server1" and reg.1.server.2.address = "server2", the phone will initially register with Server1 as the working server.
Technical Bulletin SoundPoint ® IP, SIP 2.1 2. Do not use OutBoundProxy configurations on the phone if the OutBoundProxy could be unreachable when the fallback occurs. SoundPoint IP phones can only be configured with one OutBoundProxy per registration and all traffic for that registration will be routed through this proxy for all servers attached to that registration. If Server 2 is not accessible through the configured proxy, call signaling with Server 2 will fail. 3.
Technical Bulletin SoundPoint ® IP, SIP 2.1 Registration in Per-Phone Configuration File Per-registration configuration is supported. The attribute now includes: Attribute Permitted Values Default Interpretation reg.x.server.y.lcs 0, 1 0 This attribute overrides the reg.x.lcs . If set to 1, the Microsoft Live Communications Server is supported for registration x. References 1. SIP 2.0 Administrator’s Guide for the SoundPoint IP and SoundStation IP Phones, August 2006.
