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August 7, 2000
This document lists the known problems in the Cisco SIP IP phone 7960 Version 1.0 release and contains information about the Cisco SIP IP phone 7960 (hereafter referred to as the Cisco SIP IP phone) that was not included in the Cisco SIP IP Phone 7960 Administrator Guide.
Sections in this document include the following:
The Cisco SIP IP phone 7960 firmware is available via CCO only. Therefore, to obtain a printed copy of the Cisco SIP IP Phone 7960 Administrator Guide, you must either download the PDF file of the manual from CCO or order a printed and bound copy of the manual through Cisco MarketPlace.
To obtain a PDF file of the manual which you can download and print, go to: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/admin.pdf
To order a printed and bound copy of the manual, go to Cisco MarketPlace and specify 78-10497-01.
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Note To place an order for the manual through Cisco MarketPlace, you must be a registered CCO user. |
In addition to this release note, use the following publications to learn how to install and use the Cisco SIP IP phone:
This section lists the currently known problems in the Cisco SIP IP phone 7960 Version 1.0 release.
Problem: Cisco SIP IP phone cannot parse absolute time in Expire headers (CSCdr09775)
Problem: The Cisco SIP IP phone uses half-duplex on a switch port that has been configured as full-duplex (CSCdr22751)
Problem: With a forked request, the Cisco SIP IP phone does not handle additional SIP 200 OK responses correctly (CSCdr56133)
Problem: The Cisco SIP IP phone cannot parse messages containing an o= SDP field greater than 2147483647 (CSCdr91126)
Problem: SIP REGISTER messages do not have support for SIP 3xx Redirection responses (CSCdr93802)
Problem: Daisy-chaining IP phones can cause problems
Problem: Workstation NICs cannot receive power through the network connection
Problem: LCD display issues occurring with certain types of building lighting
Problem: Powering the IP phone from multiple sources causes different results
This section contains information that has been amended or was not included in the Cisco SIP IP Phone 7960 Administrator Guide, the Cisco IP Phone 7960 Getting Started Guide, or the Cisco IP Phone 7960 Quick Reference Card. When applicable, the headings in this section correspond with the section titles in the documentation.
This section contains the following new or amended information:
In addition to the features listed in the "Supported Features" section of the "Product Overview" chapter of the Cisco SIP IP Phone 7960 Administrator Guide, the Cisco SIP IP phone also supports network call forwarding.
Network call forwarding allows the Cisco SIP IP phone user to request forwarding service from the network (via a third party tool that enables this feature to be configured). When the call is placed to the user's phone, it is redirected to the appropriate forward destination by the SIP proxy.
In addition to the inline power sources listed in the "Connecting to Power" section of the "Product Overview" chapter of the Cisco SIP IP Phone 7960 Administrator Guide, the Cisco SIP IP phone can also obtain inline power from the following sources:
The compliance information for the following SIP responses has been changed from that documented in the Cisco SIP IP Phone 7960 Administrator Guide. The changed compliance information is as follows:
| Response | Supported? | Comments |
|---|---|---|
100 Trying | Yes | The Cisco SIP IP phone generates this response for an incoming INVITE. Upon receiving this response, the phone waits for a 180 Ringing, 183 Session progress, or 200 OK response. |
183 Session progress | See comments | The SIP IP phone does not generate this message. Upon receiving this response, the phone provides early media cut through and then waits for a 200 OK response. |
302 Moved temporarily | The Cisco SIP IP phone does not generate this response at this time. Upon receiving this response, the phone sends an INVITE containing the contact information received in the 302 Moved temporarily response. | |
404 Not found | Yes | The Cisco SIP IP phone generates this response if it is unable to locate the callee. Upon receiving this response, the phone notifies the user. |
486 Busy here | Yes | The Cisco SIP IP phone generates this response if the called party is off hook and the call cannot be presented as a call waiting call. Upon receiving this response, the phone notifies the user and generates a busy tone. |
The documentation for the connection information (c=) SDP field in the INVITE messages of the following call flows in the Cisco SIP IP Phone 7960 Administrator Guide has been changed:
As illustrated in Figure 1, to place the call on hold, IN IP4 0.0.0.0 is inserted into the c= SDP field of the INVITE message in step 5.
To reestablish the call between user A and user B, the IP address of User B is inserted into the c= SDP field of the INVITE message in Step 8. This reestablishes the 2-way RTP voice path between user A and user B.
As illustrated in Figure 2, to reestablish the call between user A and user B, the IP address of User B is inserted into the c= SDP field of the INVITE message in Step 14. This reestablishes the 2-way RTP voice path between user A and user B.
As illustrated in Figure 3, to reestablish the call between user A and user B, the IP address of User B is inserted into the c= SDP field of the INVITE message in Step 15. This reestablishes the 2-way RTP voice path between user A and user B.
When using the Cisco SIP IP phone, a variety of icons can display on the phone's LCD. Table 1 lists and describes each icon that you might see while using the Cisco SIP IP phone.
| Icon | Meaning |
|---|---|
| The Cisco IP phone 7960 that you are using is running SIP. |
| The line is configured for E.164 number dialing and you can enter only numbers when placing the call. The character "x" displayed to the right of the icon indicates that registration has failed. |
| The line is configured for E.164 number dialing and ready for you to place the call. When a line is configured for E.164 number dialing, you can enter only numbers when placing the call. You can change to URL dialing at any time while dialing on a line by pressing the more soft key and then the URL soft key. The character "x" displayed to the right of the icon indicates that registration has failed. |
| The line is configured for URL dialing and you can enter both numbers and letters when placing the call. The character "x" displayed to the right of the icon indicates that registration has failed. |
| The line is configured for URL dialing and ready for you to place the call. When a line is configured for URL dialing, you can enter both numbers and letters when placing the call. You can change to E.164 number dialing at any time while dialing on a line by pressing the more soft key and then the Number soft key. The character "x" displayed to the right of the icon indicates that registration has failed. |
| The Cisco SIP IP phone configuration mode is locked. When the phone is locked, the phone's network or SIP settings cannot be modified. |
| The Cisco SIP IP phone configure mode is unlocked. When the phone is unlocked, the phone's network or SIP settings can be modified. |
The following features, while noted as SIP features in the Cisco IP Phone 7960 Getting Started Guide and the Cisco IP Phone 7960 Quick Reference Card that shipped with the phone, are not supported in the Version 1.0 of the Cisco SIP IP phone:
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In North America, TAC can be reached at 800 553-2447 or 408 526-7209. For other telephone numbers and TAC e-mail addresses worldwide, consult the following web site: http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml.
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This document is to be used in conjunction with the Cisco SIP IP Phone 7960 Administrator Guide publication.
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Posted: Mon Aug 7 13:14:46 PDT 2000
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