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These release notes describe newly incorporated features and resolved and open issues for the Cisco Analog Telephone Adaptor (ATA) 186, Release 2.13.
These release notes provide the following information:
Before you can take advantage of the features of Release 2.13, you must first download and upgrade the Cisco ATA 186 software. You can download the software at:
http://www.cisco.com/cgi-bin/tablebuild.pl/ata186
For more information, see the Cisco ATA 186 Installation and Configuration Guide.
This section contains information on new features for Release 2.13 of the Cisco ATA 186.
Release 2.13 allows the Cisco ATA 186 to register as a single endpoint with two E.164 numbers and one H.323 ID as its alias names. The endpoint type is still "terminal."
Configure LoginID0 and LoginID1 identically. This value becomes the H.323 ID. Additionally, set UseLoginID to 1. If authentication is required, configure PWD0 and PWD1 identically. UID0 and UID1 still represent the E164 numbers for line 1 and line 2, respectively, and should have different values.
When in single terminal registration mode, the Cisco ATA 186 performs the following actions when it receives an update request (URQ) message from the gatekeeper:
1. Closes all the active calls with this endpoint.
2. Replies with the update confirmation (UCF) message to the gatekeeper.
3. If UseTftp is set to 1, updates its profile from the TFTP server and reset. If UseTftp is set to 0, it simply begins another discovery and registration cycle.
The called Cisco ATA 186 now provides IP ringback across the network so it resembles the traditional Public Switched Telephone Network (PSTN). Previously the Cisco ATA 186 used only local ringback. If the IP Ringback feature is enabled, a progress indicator (PI) with progress descriptor #8 (Inband information or an appropriate pattern) is sent in the H.225 alert message in response to an H.225 Setup message. Once media-ready status is determined, ringback is generated from the Cisco ATA 186 to the caller's side.
You can now configure the calling Cisco ATA 186 to decode RTP packets (for example, tones and announcements) received prior to connection only when a valid PI is received.
To enable IP Ringback on the Cisco ATA 186, set the NetworkRingback bit (bit 19) in the configuration ConnectMode to 1. (ConnectMode = 0x00080000)
To enable the Cisco ATA 186 to require a PI before decoding RTP packets, set bit 6 in the configuration ConnectMode to 1. (ConnectMode = 0x00000040)
The Cisco ATA 186 can now negotiate four types of G.729. The codecs are G.729, G.729A, G.729AB, and G.729B. Prior to the introduction of this feature, the Cisco ATA 186 only advertised the ability to use G.729.
Cisco uses "G.729" to refer to the high-complexity G.729 as well as the low-complexity G.729A. Additionally, Cisco uses "G.729 with silent suppression" to refer to the high-complexity G.729B as well as the low-complexity G.729AB.
If you prefer G.729, configure the Cisco ATA 186 as follows:
The codecs advertised in order of preference in this case are G.729A, G.729, G.729AB, and G.729B.
If you prefer G729 with silent suppression, configure the Cisco ATA 186 as follows:
The codecs advertised in order of preference in this case are G729AB, G729B, G729A, and G729.
This section lists the issues in previous releases of the Cisco ATA 186 that are resolved in Release 2.13.
During stress test, using G729A with silence suppression off and connect mode set to IP Ringback with fast start would randomly cause audio drop. This issue has been resolved.
The Cisco ATA 186 could infrequently be led into a hanging state due to the timing of its handling of RTP packets relative to the processing of ReleaseComplete messages. This issue has been resolved.
When the Cisco ATA 186 called a Cisco 7960 IP Phone or a Cisco IOS gateway directly through CallManger or NetMeeting while sending out-of-band DTMF by using the H.245 Alphanumeric method, only one side would hear audio. This issue has been resolved.
G.729 negotiation between previous versions of the Cisco ATA 186 software and the current version resulted in one-way audio. The Cisco ATA 186 version 2.12 software tried to invoke an OpenLogicalChannel request when using G.729A even if the remote end only supported G.729, leading to a codec negotiation failure and one-way audio between platforms. This issue has been resolved.
When a call failed due to ARQ retry timeout on the caller's side, there was no tone indicator. Replacing the connected telephone handset did not put the line back to the idle state. The workaround was to press the hookflash twice quickly to return the line to the normal idle state with dial tone. This issue has been resolved.
It was possible to complete a direct IP call through a Cisco ATA 186 registered to a security gatekeeper. When the Cisco ATA 186 (configured with admission level security) was called, it generated a Cisco Access Token (CAT) and included it in the ARQ to the gatekeeper, even if the Setup message from the originating gateway did not include a CAT. This resulted in the completion of calls that should have been rejected. This issue has been resolved.
When equipped with the Swedish specification for ring impedance capacitive loading (1400 Ohms + 100 µF), the Cisco ATA 186 would stop ringing after the end of the first ring. This issue has been resolved.
A Cisco ATA 186 using version 2.12 software with FastStart and the G.729 codec would sometimes experience hanged or reset calls when calling a Cisco ATA 186 using 2.11 or earlier software versions. The problem was due to an issue in the receiving buffer that caused overflow when an incoming Setup message was too large. This issue has been resolved.
When using SIP, the codec did not change from the G.723 codec to the G.711 codec when going into conference mode while the second called party was still ringing. This issue has been resolved.
In SIP mode, the Cisco ATA 186 did not respond with a 487 message for the original INVITE message after having received a CANCEL message for it. This issue has been resolved.
In a call transfer initiated by the Cisco ATA 186 to transfer a callee to another party, the Also: header value was not in the BYE command sent to the callee. This caused the transfer to fail. This happened only when a SIP proxy server was used to authenticate the BYE command; it did not occur if the SIP proxy server did not authenticate the BYE command. This issue has been resolved.
When the Cisco ATA 186 is operating in SIP mode with the "CallCommand" entry "Uh;GQ;" replaced with "Uh;GQf;AF;", conferencing failed when the initiating caller performed a hookflash to begin the conference before the third conference member answered the call. This issue has been resolved.
This section lists open issues for Cisco ATA 186 Release 2.13.
Spurious DTMF takes place during conversation between Cisco ATA 186s.
During a call between Cisco ATA 186s, spurious out-of-band DTMF tones can be heard.
The audio breaks up after a long period during reliability testing.
A call made between two Cisco ATA 186s in G.729 mode can experience audio breakup during multiple-hour reliability tests.
A Cisco ATA 186 registered to a gatekeeper rejects calls placed from a gateway to the second port.
If the Cisco ATA 186 is registered to a gatekeeper and it is called directly by a gateway or terminal (NetMeeting) using G.723.1 and G.711u-law codecs, the call to the second port always fails. If the call is to the first port, it is successful. If the Cisco ATA 186 receives calls via a gatekeeper, calls to both ports are successful. If the Cisco ATA 186 is configured to not register to a gatekeeper and it receives calls from a gateway or a terminal, calls to both ports are successful.
1. Do not configure the Cisco ATA 186 to register a gatekeeper (set GkorProxy field to 0) or make calls to a registered Cisco ATA 186 via a gatekeeper.
2. In this software version, a single registration mode is added that allows the Cisco ATA 186 to register with a gatekeeper as a single endpoint. This allows the Cisco ATA 186 to register with a gatekeeper while the second port can also be called directly by a gateway or endpoint. To activate this mode, set LoginID0 and LoginID1 to the same H323ID and set UseLoginID to 1. The Cisco ATA 186 registers as a single endpoint with the alias names UID0, UID1, and H323ID.
The message waiting indicator (MWI) tone is lost.
In SIP mode the following sequence of actions results in a lost MWI tone:
3. Phone 1 leaves a message for Phone 2.
5. Phone 2 goes off-hook and hears the MWI tone.
6. Phone 2 goes on-hook again without checking the message.
7. Phone 2 goes off-hook again; this time MWI tone can no longer be heard.
After the caller ID feature is disabled, Phone 2 still shows the ID during call transfer.
In SIP mode with Caller ID disabled (set "CallFeatures" to 0xfff7fff7 to disable Caller ID on Phone 1 and Phone 2), direct calls correctly result in the "Anonymous" message being displayed on the Caller ID device of the callee. However, when Phone 1 of the Cisco ATA 186 transfers the Phone 2 to a third party, the callee sees the Caller ID information of the transferred party.
The Cisco ATA 186 can experience problems when sending the REGISTER message after a power reset.
After an initial successful registration with a proxy server, the Cisco ATA 186 can experience problems when sending the REGISTER message after a power reset (and subsequent DHCP, TFTP, and DNS/proxy server sequence). Debug reports from the Cisco ATA 186 show that the REGISTER message was sent and failed; however, a network analyzer shows that no message was actually sent.
Calls incorrectly branch back to the caller.
In SIP mode, if a caller on Phone 1 leaves a message for a caller on Phone 2, and within 5 seconds the callee attempts to retrieve message, Phone 2 will incorrectly dial Phone 1 instead of getting the voice prompt from the message center.
Wait for 10 seconds or more before making a call to retrieve a message.
This condition occurs when operating the Cisco ATA 186 in FAX passthrough or FAX mode at a transmission rate of 14.4Kbps or V.17 with a Cisco IOS-based gateway.
Cisco ATA 186s with serial numbers higher than ACT05280000 allow successful FAX transmission at these speeds.
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