You need the latest version of both the SIP software and bootROM to do it. Auto-answer could be configured only using provisioning. To prepare configuration files you have to do following steps:

1. In the 'sip.cfg' file, look for the line with these variables:

   <alertInfo voIpProt.SIP.alertinfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class="3"...>

   Polycom calls up class 3 in sip.cfg or ipmid.cfg file.

2. In 'sip.cfg' my ring class 'AUTO_ANSWER' looks like this:

  <ringType se.rt.enabled="1" se.rt.modification.enabled="1">

  <DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="2" se.rt.1.callWait="6" se.rt.1.mod="1"/>

  <VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/>

  <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/>

  <RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>

  <INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1"/>

  <EXTERNAL se.rt.6.name="External" se.rt.6.type="ring" se.rt.6.ringer="2" se.rt.6.callWait="6" se.rt.6.mod="1"/>

  <EMERGENCY se.rt.7.name="Emergency" se.rt.7.type="ring" se.rt.7.ringer="2" se.rt.7.callWait="6" se.rt.7.mod="1"/>

  <CUSTOM_1 se.rt.8.name="Custom 1" se.rt.8.type="ring" se.rt.8.ringer="5" se.rt.8.callWait="7" se.rt.8.mod="1"/>

  <CUSTOM_2 se.rt.9.name="Custom 2" se.rt.9.type="ring" se.rt.9.ringer="7" se.rt.9.callWait="7" se.rt.9.mod="1"/>

  <CUSTOM_3 se.rt.10.name="Custom 3" se.rt.10.type="ring" se.rt.10.ringer="9" se.rt.10.callWait="7" se.rt.10.mod="1"/>

  <CUSTOM_4 se.rt.11.name="Custom 4" se.rt.11.type="ring" se.rt.11.ringer="11" se.rt.11.callWait="7" se.rt.11.mod="1"/>

  </ringType>

  'se.rt.3.type="ANSWER"' sets Polycom phone ring type, in this case an answer, that means that phone will automatically answer without ringing.

3. Update modified files to provisioning server

4. Reload PBXware if used as provisioning server

5. Restart phone

6.

The bootROM on the phone performs the provisioning functions of downloading the bootROM, the <Ethernet     address>.cfg file, and the SIP application and uploading log files. The SIP application performs the provisioning functions of downloading all other configuration files, uploading and downloading the configuration override file and user directory, downloading the dictionary and uploading log files.

The protocol which will be used to transfer files from the boot server depends on sev-eral factors including the phone model and whether the bootROM or SIP application stage of provisioning is in progress. TFTP and FTP are supported by all SoundPoint® and SoundStation® phones. The SoundPoint® IP 301, 501, 600 and 601 and Sound- Station® IP 4000 bootROM also supports HTTP while the SIP application supports HTTP1 and HTTPS. If an unsupported protocol is specified, this may result in unex-pected behavior, see the table for details of which protocol the phone will use. The “Specified Protocol” listed in the table can be selected in the Server Type field or the Server Address can include a transfer protocol, for example http://usr:pwd@server (see 2.2.1.3.3 Server Menu on page 10). The boot server address can also be obtained via DHCP. Configuration file names in the <Ethernet address>.cfg file can include a transfer protocol, for example https://usr:pwd@server/dir/file.cfg. If a user name and password are specified as part of the server address or file name, they will be used only if the server supports them.

NOTE: A URL should contain forward slashes instead of back slashes and should not contain spaces. Escape characters are not supported. If a user name and password are not specified, the Server User and Server Password will be used.

For downloading the bootROM and application images to the phone, the secure HTTPS protocol is not available. To guarantee software integrity, the bootROM will only download signed bootROM or application images. For HTTPS, widely recog-nized certificate authorities are trusted by the phone and custom certificates can be added. See 6.1 Trusted Certificate Authority List on page 151. Using HTTPS requires that SNTP be functional. Provisioning of configuration files is done by the application instead of the bootROM and this transfer can use a secure protocol.

Configuring Cisco phones for speakerphone page.

Note that Cisco phones are not supporting special sip header for auto-answer, that means that You can’t configure these phones to auto answer only to speaker phone page and to ring for ordinary incoming calls. You can only configure one line who will be set to auto answer any call not just page call. That means that with Cisco you have to have one special separated line for speakerphone page.

To set already configured line on Cisco phone you have to do following steps:

1. Go into menu 'Settings -> Call Preferences -> Auto Answer (intercom)'

2. Set a new line you've just created as auto-answer by choosing 'Line x ON/OFF'

NOTE: Paging tested with Firmware version 1.6.5.0043