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http://192.168.2.170/refresh
If you are using TFTP when the Cisco ATA 186 is plugged in, the Cisco ATA 186 will try to contact the TFTP server to download its configuration. This method is not secure unless you are using EncryptKey. See the Encrypt Key section on page 3-8.
cfgfmt.exe and ptag.dat Files
Bundled with the Cisco ATA 186 software is the program cfgfmt.exe and the file ptag.dat. These should be placed the directory used to store the files for transfer using TFTP. The cfgfmt program is used to convert a text-based user profile for
Configuring the Cisco ATA 186 About Profile and Configuration Security
the Cisco ATA 186 to a binary file sent by the TFTP server to the Cisco ATA 186 to update its configuration parameters. The cfgfmt.exe program is used with the following syntax: cfgfmt [-eRC4Password] [-tPTagFile] input output
eRC4Password is the optional RC4 key to encrypt the binary TFTP file provided by the cfgfmt program tPTagFile is the optional command used to specify a ptag file other than the one provided (ptag.dat) input is the name of the text-based profile of the Cisco ATA 186 that will be converted to a TFTP binary file output is the name of the TFTP binary file produced by the cfgfmt program
Updating the Profile from the TFTP Server
To update the Cisco ATA 186 profile from the TFTP server before the CFGINTERVAL expires, open your web browser and enter: http://ipaddress/refresh where ipaddress is the IP address of the Cisco ATA 186 you want to update. The Cisco ATA 186 responds with an ok page if idle; otherwise, it responds with a later page. If you have physical access to the Cisco ATA 186, you can power cycle the Cisco ATA 186 to update the profile from TFTP server.
About Profile and Configuration Security
This section includes information on passwords and other security methods.

Passwords

To password-protect your Cisco ATA 186:
Chapter 3 About Using the DHCP Server
Set the UIPassword parameter to a numeric password by using the web server interface or TFTP profiling. You will be prompted for a password when you try to access the web server or a configurable IVR parameter.
In web server mode, enter the password in the UIPassword field of the password challenge page. In IVR mode, enter the password, followed by the # key, at the p-a-s-s-w-d prompt.

Encrypt Key

Encrypt Key encrypts binary files being transferred over TFTP. You can change this key for each Cisco ATA 186, so that only one particular box can decode the information. You can change the encrypt key, using the IVR or web interface. See the cfgfmt.exe and ptag.dat Files section for more information. The Cisco ATA 186 polls the server at intervals set in CfgInterval to see if it needs to be upgraded. You can customize this service. For example, you can route all calls from a particular Cisco ATA 186, based on the Gatekeeper ID, to an operator.

About Using the DHCP Server
DHCP option 60, DHCP_VENDOR_CLASS_ID, is set to the value ATA186 so that the DHCP server can identify a Cisco ATA 186. Parameters that you can set using DHCP are:
Client IP address Client Subnet maskDHCP option 1 Routers on the client's subnetDHCP option 3 Domain name serversDHCP option 6 (The Cisco ATA 186 takes up to two DNS servers)
Configuring the Cisco ATA 186 About Using the DHCP Server
Network time protocol (NTP) serversDHCP option 42 (The Cisco ATA 186 takes up to two NTP servers)
TFTP server nameDHCP option 66
DNS, TFTP, and NTP servers can be overwritten by the value of the corresponding parameters in the local box profile (for example, the DNS1IP, DNS2IP, TftpURL, and NTPIP parameters). If you are not using DHCP, you must manually enter the IP address, network route address, and subnet mask.
Chapter 3 Configuring Codec Options
Configuring Codec Options
You can configure the various Codec call options for use with the Cisco ATA 186.
The Cisco ATA 186 can support two simultaneous G.723 calls or one G.729A call. When using G.729A, the second line must use G.711 u-law or a-law. The default voice codec is G.723.
To select G.723 as the preferred low-bit-rate codec (LBRCodec) for receive and transmit modes, enter 0 into the LBCodec field on the web page. To select G.729A, enter 3. To select G.723 as the preferred receive codec (RxCodec) enter 0 into the RxCodec field on the web page. To select G.729A, enter 3. To select G.723 as the preferred transmit codec (TxCodec), enter 0 into the TxCodec field on the web page. To select G.729A, press 3.

Step 2 Step 3

Protocol-Specific Configurations
This chapter contains information on selecting protocols and services for your system.
About Signaling Protocols
You can select either H.323 or Session Initiation Protocol (SIP) as the operating signaling protocol for the Cisco ATA 186. Both signaling protocols offer optional network control servers. With H.323, pre-call and call control services are offered by a gatekeeper, while in SIP, a proxy server can receive call transaction requests and return responses on behalf of the Cisco ATA 186. Some parameters and supplementary services are available with SIP, some are available only with H.323, and others are available with both protocols.

About H.323-Specific Configurations
The Cisco ATA 186 uses ITU H.323, Version 2 as the default signaling protocol. When operating in H.323 mode, the Cisco ATA 186 registers with a gatekeeper to handle call control services. A full registration request (RRQ) is performed at power-up. In order to let the gatekeeper know it is still on the network, the Cisco ATA 186 periodically refreshes this registration by sending an abbreviated RRQ. The value of the GKTimeToLive configuration parameter determines the period between refreshes, in seconds.
Chapter 4 About Gatekeeper Requirements for H.323
To use the H.323 security features, you must specify the level of authentication by means of the AutMethod configuration parameter. The settings are as follows:
0no authentication 1Cisco registration level 2Cisco admission level
Make sure these levels are also enabled on the gatekeeper and gateway.
About Gatekeeper Requirements for H.323
The gatekeeper must meet these requirements:
It must be H.323- or SIP-complaint. It must run the applicable version of Cisco IOS software for the features and protocol you want to use. It must support H.323 or SIP, but only one at a time.
No specific configuration is required; configure the gatekeeper as you would for any IP phone or Voice over IP (VoIP) configuration. The default configuration is IP routing off.

Enabling IP Routing

To enable IP routing so that you can run Cisco IOS software, enter: ip routing

Chapter 4

Protocol-Specific Configurations About Gatekeeper Requirements for H.323
Connecting to a Network Time Protocol Server
If you want to use Caller ID (SIP) or security features (H.323), connect the gateway to a functioning network time protocol (NTP) server. When using Cisco IOS, enter: ntp server ip_address clock timezone PST -8
For information on how to access accounting information, see your Cisco IOS documentation.

Using ISDN/EI

If you are using ISDN, the requirements for E1 are:
ISDN switch-type primary-5ess voice rtp send-rcv
Configuration depends on your WAN connection. The following example shows a 24-channel PRI connected to a T1 VIC slot:
controller Tframing esf clock source linelinecode b8zs pri-group timeslots 1-24

Dial the first number. When the person you called answers, perform a hook flash; that is, hang up quickly and pick up the phone quickly. This will put the first person you called on hold and you will hear a dial tone.

Chapter 5

Configuring Supplementary Services Common Supplementary Services
Dial the second person and speak normally when that person answers. Perform a second hook flash and press 2 on the telephone handset to get back to the first person. You can continue to switch back and forth between the two callers. To conference with both callers at the same time, perform a hook flash and press 3 on the telephone handset. Once you conference all three callers, the only way to drop a caller is for that caller to hang up. (Optional) To conference in additional callers, the last person called with a Cisco ATA 186 can call an additional person, and so on. This is known as daisy-chaining.

Step 6

About Call Waiting
This section describes the Call Waiting feature.

Call Waiting in the U.S.

If someone calls you while you are speaking on the telephone, you can answer by performing a hook flash. You cannot conference in all 3 callers, but the first person you called could call someone else and daisy-chain them in. If there is no answer after one minute, the caller will hear 3 beeps and a busy signal. When the Cisco ATA 186 is configured to use Call Waiting by default, press *70 to disable Call Waiting for the duration of the next call.

Call Waiting in Sweden

If someone calls you while you are speaking on the telephone, you can answer by performing a hook flash and pressing 2 to answer or 3 to conference them with the person you are already speaking to. You can also press 3 later during the call to conference. Performing a hook flash and pressing 1 hangs up the first caller and answers the second call. If there is no answer after one minute, the caller will hear 3 beeps and a busy signal.
To enable Call Waiting for Sweden, press *43#. When the Cisco ATA 186 is configured to use Call Waiting by default, press #43# to disable Call Waiting for the duration of the next call.

About Call Forwarding

In H.323, it is necessary to have a gatekeeper capable of handling call forwarding and call return supplementary services. In SIP, the Cisco ATA 186 handles these services. There are 3 types of call forwarding.
Forward Unconditionalforwards every call that comes in. Forward When Busyforwards calls when the line is busy. Forward on No Answerforwards calls when the telephone is not answered after the configured period of 0-63 seconds.
You can activate only one of these services at a time.
Forwarding Calls in the U.S.

TFTP URL Timezone ToConfig 0=provisioned 1=new (default) Trace Flags 0=disable (default) 1=enable Tx Codec 0=G.723.1 (default) 1=G.711A 2=G.711 3=G.729A UDP TOS Bits UID 1 Upgrade Software Upgrade Language to English
80001 See Appendix B, Parameters and Defaults
Enable logging of SIP messages.
Select the audio codec type to use to encode data for transmission. The audio codec type for receiving does not have to be the same as the audio codec type for transmitting.
Change or upgrade the voice prompt language to English. See Chapter 6, Upgrading the Cisco ATA 186 Software for more information. Enter the User ID (telephone number) for the Phone 2 port.

Use Login ID

Option Use TFTP 0=disable (default) 1=enable UID 0 Version Number

Item Number 305

Enter the User ID (telephone number) for the Phone 1 port. Listen to the version number of the Cisco ATA 186 software.

Table A-2

Voice Menu Options in Numerical Order
Option IP Address Network Route Address UID 0 PWD 0 Gatekeeper/Proxy Server IP Address Alternate Gatekeeper (Altgk) Subnet Mask
Description Enter the IP address. This number can be automatically assigned when DHCP is enabled. Enter the network route address. This number can be automatically assigned when DHCP is enabled. Enter the User ID (telephone number) for the Phone 1 port. Enter the password associated with the gateway account. Enter the gatekeeper or proxy server IP address. (H.323) Enter the subnet mask. This number can be automatically assigned when DHCP is enabled. The default is 255.255.255.0. Enter the IP address of the gateway. This number can be automatically assigned when DHCP is enabled.

Gateway IP Address

Option UID 1 PWD 1 Dynamic Host Configuration Protocol (DHCP) 0=Disable 1=Enable (Default)
Description Enter the User ID (telephone number) for the Phone 2 port. Enter the password associated with the second gatekeeper account. This command controls whether the Cisco ATA 186 can automatically obtain configuration parameters from a server over the network. Listen to the IP address. Listen to the IP address of the network route. Listen to the subnet mask. Listen to the media access control (MAC) address. Select the number of frames per packet to be transmitted when using the audio codec G.723.1 or G.729A. Select the audio codec type to use to decode received data. The call-receiving station automatically adjusts to the call-initiating station's audio codec type if the call-receiving station supports that audio codec.
Review IP Address Review Network Route IP Address Review Subnet Mask Review MAC Address Num Tx Frames Values: 1 through 12. Default=2

Table B-3 Common Operating Parameters and Defaults

Parameter AutMethod

Value Type Bitmap
IVR Access Code Default 0x00000000
Authentication method. Possible values 92 are:
bit 0-1 (mask 0x3): 0 for no authentication, 1 for authentication Cisco registration level security (H.235), 2 for Cisco Systems admission level security. bit 2 (mask 0x4): prefix password field when registering. U.S. Cmd Table

CallCmd

N/A Controls call commands such as turning on and off Caller ID and so on. (248 characters maximum.) U.S. Cmd Table CallCmd:Af;AH;BS;NA;CS;NA;Df;E B;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67; gA*82;fA90v;OI;H72v;bA74v;cA75v; dA73;eA*67;gA*82;fA*70;iA*69;DA *99;xA;Uh;GQ; Swedish Cmd Table CallCmd:BS;NA;CS;NA;Df;EB;Ff0;A Rf1;HPf2;EPf3;AP;Kf1;HFf2;EFf3;A Ff4;HQ;Jf1;HFf2;EFf3;AFf4;HQ;Af4; HQ;I*31;gA31;gA*90*v;OI;H*21*v;b A*61*v;dA*67*v;cA21;eA61;eA67;e A*31;gA31;gA*43;hA43;iA*69;DA*9 9;xA;Uh;GQ;
Appendix B Operating Parameters

Table B-3

Common Operating Parameters and Defaults (continued)

Parameter CallerIdMethod

Value Type Integer
Description CallerID/DTMFMethod. Possible values are: bit 0-1 (mask 0x3): method; 0=Bellcore, 1=DTMF, 2=ETSI, 3=FSK (0) bit 2 (mask 0x4): method type; 0=type 1, 1=type 2 (0) if(method=0) { bit 3-8 (mask 0x1f8): max no. of digits (12) bit 9-14 (mask 0x7e00): max no. of chars (15) bit 15-20 (mask 0x1f8000): special chars (3) } else { bit 3-6 (mask 0x78): start digit (12) bit 7-10 (mask 0x780): end digit (14) bit 11 (mask 0x800): polarity reversal (1) bit 12-16 (mask 0x1f000): max no. of digits (15) }
Sweden=0x0ff61 Denmark=0x0fde1 USA=0x19e60
IVR Access Code Default 316 0x00019e60
Parameter CallWaitCallerId
Description Caller ID on CallWaiting. Possible values are:
IVR Access Code Default 317 0x003c33d0
bit 0-5 (mask 0x3f): max no of digits (16) bit 6-11 (mask 0xfc0): max no of characters (15) bit 12-17 (mask 0x3f000): special chars (3) bit 18-21 (mask 0x3c0000): ack digit (15)

Parameter ConnectMode

IVR Access Code Default 0x00060000
Connection mode of the protocol used. 311 Possible values are:
bit 0 (mask 0x1): 0 for slow start and 1 for fast start (h323). bit 1 (mask 0x2): 1 use h245 tunneling. bit 3 (mask 0x8): 1 means alternate gatekeeper need register. bit 12-8 (mask 0x1f00): offset to payload 96 (0-23). bit 13 (mask 0x2000): 0 use G.711ulaw; 1 use G.711alaw. bit 14 (mask 0x4000): 0 use fax pass through and 1 use codec negotiation in sending fax. bit 15 (mask 0x8000): 0 means enable/1 means disable detecting fax pass through. bit 16 (mask 0x10000): 1 enables SIP to remove the registration before adding a new one. bit 17 (mask 0x20000): 1 enables SIP call forwarding to be performed by the Cisco ATA 186. bit 18 (mask 0x40000): 1 enables SIP call return performed by the Cisco ATA 186.

Example 2 (Default Dial Plan)
*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
consists of the following rules: *St4-If the first digit entered is *, all other dial plan rules are voided. Additional digits can be entered after the initial * digit, and the timeout before automatic dial string send is 4 seconds.
#St4Same as above, except with # as the initial digit entered. 911If the dial string 911 is entered, send it immediately. 1>#t8.r9t2If the first digit entered is 1, the timeout before automatic send is 8 seconds. The terminating character # can be entered at any time to manually send the dial string. After the 11th digit is entered, the timeout before automatic send changes to 2 seconds. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character. 0>#t811.rat4If the first digit entered is 0, the timeout before automatic send is 8 seconds, and the terminating character # can be entered at any time to manually send the dial string. If the first 3 digits entered are 011, then after an additional 11 digits are entered, the timeout before automatic send changes to 4 seconds. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character. ^1t4>#.If the first digit entered is anything other than 1, the timeout before automatic send is 4 seconds. The terminating character # can be entered at any time to manually send the dial string. The user can enter more digits until the dial string is sent by the timeout or by the user entering the # character.
Dial Plan Parameters and Defaults Dial Plan Blocking

Dial Plan Blocking

Dial Plan blocking prevents most invalid dialed digits from being sent. You can change the default interdigit timeout of 9 seconds by adding the following rule to your dial plan string: In where n is 1-9 or a-z (for 10-35) seconds. For example, enter an interdigit timeout of 12 seconds as:
Ic|[the rest of your dial plan rules]
Specifying your own interdigit timeout also changes the behavior of the dial plan so that, rather than the entire dial string being sent at timeout, it is sent only as a result of a matching rule or time intended by a matching rule.
Appendix D Dial Plan Blocking
Paid Services and Call Features Parameters and Defaults
This appendix provides information on the paid services and call features parameters and defaults that you can use to provision your Cisco ATA 186. The paid services parameter is in bitmap format. The IVR access code for paid services is 314. The IVR access code for call features is 315. The default for paid services is 0xffffffff. The default for call features is 0xffffffff. Call features is a 32-bit bitmap value. The lower 16-bits are used for channel 0; the upper 16-bits are used for channel 1. The call features and paid services parameters use the same bit masks. Paid services indicates which service the user has subscribed to, while call features indicates which feature is statically enabled by the user. Not all supplementary services can be disabled by the user.

Only positive values can be provisioned to the Cisco ATA 186. For negative values, use the complement value of the 16-bit 2. For example, enter -1 as 65535 or 0xffff. freq1 is the transformed frequency of the second frequency component (-32768 to 32767). level0 is the transformed amplitude of the first frequency component (-32768 to 32767). level1is the transformed amplitude of the second frequency component (-32768 to 32767). steady controls whether the tone is constant or intermittent. A value of 1 indicates a steady tone and causes the Cisco ATA 186 to ignore the on-time and off-time parameters. A value of 0 indicates an on/off tone pattern and causes the Cisco ATA 186 to use the on-time and off-time parameters. on-time controls the length of time the tone is heard in milliseconds (ms) expressed as an integer from 1 to 0x7FFFFFFF.

Appendix F

Call Progress Tone Parameters and Defaults Playback Tone
off-time controls the length of time between audible tones in milliseconds (ms) expressed as an integer from 1 to 0x7FFFFFFF. total time controls the length of time the tone is audible in multiples of 10 milliseconds (ms) expressed as an integer from 1 to 0x7FFFFFFF. A value of 0 causes the tone to remain audible indefinitely.
Frequency ranges from 0 to 4000 (Hz) Transformed Frequency = 32767 cos (2piFrequency/8000) Transformed Amplitude = A 32768 sin (2piFrequency/8000) The scaling factor A selects the volume level of the tone. To customize the playback tone parameters, select a value for A based on the desired volume and the number of frequency components in the relevant tone. Table F-1 shows several values of A and the approximate volume level for each value of A for tones that consist of one and two components.
Example -- Calculating The Volume Levels For Dial Tone
A U.S. dial tone consists of two frequency components, 350 Hz, shown in this example as level0, and 440 Hz, shown as level1. To set the dial tone to a volume of -10dBm for a two-component tone at -10dBm, use a multiplier of 0.35. The formula for level0 uses this value of A in the Transformed Amplitude formula as follows: 0.sin (2pi 350 / 8000) = 3194 The formula for level1 is: 0.sin (2pi 440 / 8000) = 4047 The approximate values to enter for level0 and level1 are 3194 and 4047, respectively.
Appendix F Example Call Progress Tone Parameters

Table F-1

Volume Levels for a Standard U.S. 600-Ohm Impedance Board
Volume Level for One Tone Volume Level for Two Tones Scale Factor (A) 0.10 0.175 0.35 0.50 Component (dBm) -24 -19 -13 -10 Components (dBm) -21 -16 -10 -7

Appendix G

Call Commands Syntax

Context-Identifiers

Table G-1 Context Identifiers
Identifier A B C D E F G H I J K L M N O P Q R S T U V W X
Context CONFERENCE PREDIAL PREDIAL_HOLDING CONNECTED CONNECTED_HOLDING CONNECTED_ALERTING HOLDING CONFIGURING CONFIGURING_HOLDING 3WAYCALLING CALLWAITING IDLE RINGING DIALING CALLING Reserved (ANSWERING) Reserved (CANCELING) Reserved (DISCONNECTING) WAITHOOK DIALING_HOLDING CALLING_HOLDING Reserved (ANSWERING_HOLDING) Reserved (HOLDING_HOLDING) Reserved (CANCELING_HOLDING)

Table G-1

Context Identifiers (continued)

Identifier Y Z a b

Context Reserved (DISCONNECTING_HOLDING) Reserved (HOLDING_ALERTING) WAITHOOK_ALERTING WAITHOOK_HOLDING

Action Identifiers

Table G-2 Action Identifiers
Identifier A B C D E F G H I N O P Q R a b
Action NONE Seizure (User Intendes To Dial Or Configure Continue to dial Call Return Hold the active call Retrieve the waiting call Cancel the call attempt Disconnect the call Blind transfer the call to the number Go to configuration mode Release the call Answer the incoming call Transfer with consultation Say busy to the caller None Forward all calls to the given number

Table G-2

Action Identifiers (continued)

c d e f g h i x

Forward on busy to the given number Forward on no answer to the given number Cancel call forward CLIP for the next call CLIR for the next call Enable Call Waiting for the next call Disable Call Waiting for the next call Enable Fax Mode for the next call

Call Command Example

In addition to the provisioned call commands, the Cisco ATA 186 has a default list of call commands to handle common call scenarios. The default call commands can be overwritten by the provisioned call commands. If any Context/Input-Sequence appears in both the default Call Command string and the manually entered string, the manually entered value takes precedence. The default Call Command string is as follows:
Bf;BAN;CA;CN;CAf;OF;Df;EB;I@f;OF;H@f;OA;Lo;BAf;BA;Mo;PA;ND;CAf;OA;Of;G A;Pf;HA;Qf;OA;Rf;OA;Sf;OA;TD;CAf;OF;Uf;GF;Vf;HF;Wf;FF;Xf;AF;Yf;AF;Zf;A P;bf;OF;af;OP;
In this section, the Call Command string is broken down into its component Context-Command-Lists as follows:
Call Command Fragment; Context-Identifier Input-Sequence1 Action1 Action2; (optional) Input-Sequence2 Action1 Action2;
When reading a Call Command string, you can identify the end of one Context-Command-List and the beginning of the next by noting whether there is a terminating ';' immediately after the second Action-Identifier, or whether it is followed by an input sequence.

PREDIAL: Phone just went off-hook but no DTMF entered yet; Cisco ATA 186 plays dial-tone
DIALING: User entering phone number, which R: Abort dialing, restarts dial-tone, and revert to is parsed with the given dial-plan rules PREDIAL;Invalid phone number: Abort dialing, plays fast-busy, and goto WAITHOOK;

Table G-3

Call Command Behaviors (continued)
Context CONFIG: User configuring a supplementary service
Commands US *69: Call Return #72v#: Forward unconditiona to number specified in 'v'l; (PacBell use 72#)#73:Cancel any call forwarding; (PacBell use 73#)#74v#: Forward on busy to number specified in 'v'; (PacBell don't enable this service from the phone)#75v#: Forward on no answer to number specified in 'v'; (Pac Bell don't enable this service from the phone)*67: CLIR in the next call (if global profile is CLIP);*82: CLIP for the next call (if global user profile is CLIR);*70: Disable call waiting in the next call;*99: Enable Fax Mode in the next call; (non-standard) Sweden *21*v#: Forward unconditionally to number specified in 'v';*67*v#: Forward on busy to number specified in 'v';*61*v#: Forward on no answer to number specified in 'v';#21#: Cancel any call forwarding;#67#: Cancel any call forwarding;#61#: Cancel any call forwarding;#31#: CLIR in the next call;*31#: CLIR in the next call; (Note: no CLIP, maybe all calls by default is CLIP in Sweden?)*43#: Enable call waiting in the next call (Sweden allows globally disable call waiting?);#43#: Disable call waiting in the next call;*69#: Call Return; (non-standard)*99#: Enable Fax Mode in the next call; (non-standard)All RegionsR or any unrecognized sequence: Abort configuration, restart dial-tone, and revert to PREDIAL;Any complete configuration sequence: Carry out the configuration command, restart dial-tone, and revert to PREDIAL
R: Cancel the outgoing call, restarts dial-tone, and CALLING: Phone number is sent; Cisco ATA 186 is waiting for response from the revert to PREDIAL far end

Context

Commands
RINGING: Cisco ATA 186 is ringing the phone OFH: Stop ringing, answer the call, and goto to alert user of an incoming call CONNECTED CONNECTED: The Cisco ATA 186 is connected with one far end party; Cisco ATA 186 may be the caller or the callee WAITHOOK: Far end hangs up while in CONNECTED state; Cisco ATA 186 plays fast-busy after 5 seconds in this state CONNECTED_ALERTING: Cisco ATA 186 receives another call while in CONNECTED state; Cisco ATA 186 plays Call Waiting tone periodically (every 10 seconds for US; l second for Sweden) US and Sweden: R: Hold current call, play dial-tone to dial 2nd number, and goto PREDIAL_HOLDING R: Stop fast-busy, start dial-tone, and goto PREDIAL

CONFIG_HOLDING: A connected FE is placed US*67: CLIR for the next call;*82: CLIP for the next on hold, while the Cisco ATA 186 is entering a call;#90v#: Blind transfer to the number specified in configuration command 'v'; disconnect the call and goto PREDIALSweden#31# or *31#: CLIR in the next call*90*v#: Blind transfer to the number specified in 'v'; disconnect the call and goto PREDIAL (non-standard)All RegionsR or any unrecognized sequence: Abort configuration, restarts dial-tone, and goto to PREDIAL_HOLDING A complete configuration sequence: Carries out the command, and goto PREDIAL_HOLDING DIALING_HOLDING: Cisco ATA 186 user entering a second phone number to call while placing a connected call on-hold CALLING_HOLDING: Cisco ATA 186 waiting for a second far end to response while placing a connected call on-hold WAITHOOK_HOLDING: The AFE hangs-up to disconnect the current call while there is a WFE being put on-hold AITHOOK_ALERTING: The AFE hangs-up while a waiting call is alerting Collected digits match a dial-plan rule: Call the new number, and goto CALLING_HOLDING;R: Aborts dialing and revert to PREDIAL_HOLDING R: Cancel the call and revert to PREDIAL_HOLDING;(ONH: Cancel the call and transfer the waiting party to the callee, and revert back to PREDIAL) R: Retrieve the WFE and goto CONNECTED
W R: Stop CWT, answer the waiting call, and goto CONNECTED;WFE cancels the call: Stop CWT, goto WAITHOOK;ONH: Goto IDLE (in which Cisco ATA 186 automatically starts ringing the phone, and goto RINGING)

U.S. Call Command

The default U.S. Call command is: Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67;gA*82;fA#90 v#;OI;H#72v#;bA#74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;x A;Uh;GQ; Af;AH; Conference hook-flash; NONE Disconnect-the-call; BS;NA; Predial #|*; Go-to-configuration-mode NONE; CS;NA; Predial_Holding #|*; Go-to-configuration-mode NONE; Df;EB; Connected hook-flash; Hold-the-active-call Seizure; Ff;EP; Connected_Alerting hook-flash; Hold-the-active-call Answer-the-incoming-call; Kf;EFh;HQ; CallWaiting hook-flash; Hold-the-active-call Retrieve-the-waiting-call on-hook; Disconnect-the-call Transfer-with-consultation; Jf;AFh;HQ; 3WayCalling hook-flash; NONE Retrieve-the-waiting-call on-hook; Disconnect-the-call Transfer-with-consultation;
I*67;gA*82;fA#90v#;OI; Configuring_Holding *67; CLIR-for-the-next-call NONE *82; CLIP-for-the-next-call NONE #90v#; Release-the-call Blind-transfer-the-call-to-the-given-number; H#72v#;bA#74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;

Configuring

#72v#; Forward-all-calls-to-the-given-number NONE #74v#; Forward-on-busy-to-the-given-number NONE #75v#; Forward-on-no-answer-to-the-given-number NONE #73; Cancel-call-forward NONE *67; CLIR-for-the-next-call NONE *82; CLIP-for-the-next-call NONE *70; Disable-call-waiting-for-the-next-call NONE *99; Enable-fax-mode-for-the-next-call NONE; Uh;GQ; Calling_Holding on-hook; Cancel-the-call-attempt Transfer-with-consultation;;

The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.

Audience

This guide is intended for service providers and network administrators who administer Voice over IP (VoIP) services using the Cisco ATA. Most of the tasks described in this guide are not intended for end users of the Cisco ATA. Many of these tasks impact the ability of the Cisco ATA to function on the network, and require an understanding of IP networking and telephony concepts.

Preface Organization

Table 1 provides an overview of the organization of this guide.
Table 1 Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrators Guide (SIP) Organization
Chapter Chapter 1, Cisco Analog Telephone Adaptor Overview
Description Provides descriptions of hardware and software features of the Cisco ATA Analog Telephone Adaptor along with a brief overview of the Session Initiation Protocol (SIP). Provides information about installing the Cisco ATA. Provides information about configuring the Cisco ATA and the various methods for configuration. Provides information about SIP services that the Cisco ATA supports. Provides information on all parameters and defaults that you can use to configure the Cisco ATA. Provides the Cisco ATA call commands for SIP. Provides instructions for configuring both ports of the Cisco ATA to support fax transmission. Provides instructions for remotely upgrading Cisco ATA software. Provides basic testing and troubleshooting procedures for the Cisco ATA. Provides end-user information about pre-call and mid-call services. Provides a quick-reference list of the voice configuration menu options for the Cisco ATA. Provides physical specifications for the Cisco ATA. Provides Cisco ATA call flows for SIP scenarios. Provides definitions of commonly used terms. Provides reference information.
Chapter 2, Installing the Cisco ATA Chapter 3, Configuring the Cisco ATA for SIP Chapter 4, Basic and Additional SIP Services Chapter 5, Parameters and Defaults, Chapter 6, Call Commands Chapter 7, Configuring and Debugging Fax Services Chapter 8, Upgrading the Cisco ATA Signaling Image Chapter 9, Troubleshooting Appendix A, Using SIP Supplementary Services Appendix B, Voice Menu Codes Appendix C, Cisco ATA Specifications Appendix D, SIP Call Flows Glossary Index

Command Example

atapname.exe 10.20.30.40.50.60

Command Output

ata0a141e28323c
The same functionality is available from the voice configuration menu (voice menu code 84#), which will announce the Cisco ATA profile name.
Using the EncryptKey Parameter and cfgfmt Tool
The EncryptKey parameter encrypts binary files being transferred over TFTP. You can change this key for each Cisco ATA, so that only one specific Cisco ATA can decode the information. By default, the Cisco ATA-specific ata<macaddress> configuration file is not encrypted. If encryption is required, however, you must manually configure the EncryptKey parameter before you boot up the Cisco ATA so that the TFTP method is secure. Use either the voice configuration menu (see the Voice Configuration Menu section on page 3-15) or the Cisco ATA web configuration page (see the Cisco ATA Web Configuration Page section on page 3-18) to configure the EncryptKey parameter.
Because the factory-fresh ATA cannot accept encrypted configuration files, the first unencrypted file, if intercepted, can easily be read. (You would still have to know the data structure format in order to decode the binary information from the unencrypted file.) Therefore, the new encryption key in the unencrypted file can be compromised. Set the EncryptKey parameter to a nonzero value. When this value is nonzero, the Cisco ATA assumes that the binary configuration file on the TFTP server is to be encrypted with this key by means of the RC4 cipher algorithm. The Cisco ATA will use this key to decrypt the configuration file. The Cisco ATA EncryptKey parameter and the encryption key used in the cfgfmt tool command syntax must match.
For security reasons, Cisco recommends that you set the UIPassword parameter (if desired) in the configuration file and not by using one of the manual configuration methods. The cfgfmt.exe syntax affects how the EncryptKey parameter is used, as shown in the following examples. In these examples, input_text is the ata<macaddress>.txt file that you will convert to binary to create the ata<macaddress> configuration file for the Cisco ATA; output_binary is that binary ata<macaddress> file, and Secret is the encryption key.

Syntax examples

cfgfmt -tpTagFile input-text-file output-binary-file If input-text-file sets the Cisco ATA EncryptKey parameter to 0, then output-binary-file is not encrypted. If the input-text-file sets EncryptKey to a non-zero value, then output-binary-file is encrypted with that value.

NatServer

This parameter allows you to specify a server to which a dummy, single-byte UDP packet is sent to maintain a Network Address Translation (NAT) during a session. NatServer can contain up to 47 characters in fully qualified domain name (FQDN) or IP format with an optional port parameter (separated from the address by a colon); for example, xyz.cisco.com:1234. If no port is specified, the default port of 5060 is assumed.
IP address or FQDN format
Maximum number of characters: 47
5060 is the default port if no port is specified.
Parameters and Defaults Operating Parameters

NatTimer

This parameter allows you to specify a retransmission interval for sending a dummy packet to NatServer. The interval is in seconds and is specified in bits 0-11 of this parameter. The upper 20 bits are reserved and should be set to 0.

Bitmap

0, which means that no dummy packets will be sent to the NatServer.

Operating Parameters

This section describes the following parameters, which are located in the green portion of the Cisco ATA Web Configuration Page:

LBRCodec, page 5-20 AudioMode, page 5-20 RxCodec, page 5-21 TxCodec, page 5-22 NumTxFrames, page 5-22 CallFeatures, page 5-23 PaidFeatures, page 5-24 CallerIdMethod, page 5-25 FeatureTimer, page 5-26 Polarity, page 5-27 ConnectMode, page 5-28 AutMethod, page 5-30 TimeZone, page 5-30 NTPIP, page 5-30 AltNTPIP, page 5-31 DNS1IP, page 5-31 DNS2IP, page 5-31 UDPTOS, page 5-32 SigTimer, page 5-32 OpFlags, page 5-34 VLANSetting, page 5-35
Chapter 5 Operating Parameters

LBRCodec

This parameter allows you to specifiy which low-bit-rate codecs are available. The Cisco ATA is capable of supporting two G.723.1 connections or one G.729 connection. When G.723.1 is selected as the low-bit-rate codec, each FXS port is allocated with one G.723.1 connection. When G.729 is selected, only one FXS port is capable of operating with the G.729 codec. The allocation of the G.729 resource to the FXS port is dynamic. The G.729 resource, if available, is allocated to an FXS port when a call is initiated or received; the resource is released when a call is completed. The following values are valid:
0Select G.723.1 as the low-bit-rate codec. 3Select either G.729 as the low-bit-rate codec.
RxCodec, page 5-21 TxCodec, page 5-22

0 or 3

AudioMode
This parameter represents the audio operating mode. The lower 16 bits are for the Phone 1 port, and the upper 16 bits are for the Phone 2 port. Table 5-1 on page 5-21 provides definitions for each bit.

0x00150015

Table 5-1
AudioMode Parameter Bit Definitions

Bit Number 4-5: DtmfMethod
Definition 0/1Disable/enable G.711 silence suppression. 0Enable selected low-bit-rate codec in addition to G.711. This setting is the default. 1Enable G.711 only. 0/1Disable/enable fax CED tone detection. 0/1Enable/disable fax CNG tone detection. 0Always in-band (send and receive, do not send SDP info) 1By negotiation (send SDP info, enable receive, decode others SDP information, send depends on others SDP information) 2Always out-of-band (send SDP info, enable receive, decode others SDP information, always send). 3Reserved.

Reserved.

RxCodec
Use this parameter to specify receiving-audio codec preference. The following values are valid:
0G.723 (can be selected only if LBRCodec is set to 0) 1G.711A-law 2G.711-law 3G.729a (can be selected only if LBRCodec is set to 3)

TxCodec

Use this parameter to specify the transmitting-audio codec preference. The following values are valid:
0G.723 (can be selected only if LBRCodec is set to 0) 1G.711A-law 2G.711-law 3G.729A (can be selected only if LBRCodec is set to 3)

NumTxFrames

Use this parameter to select the default RTP packet side in number of frames per packet. The Cisco ATA default frame sizes are as follows:
G.711 and G.72910 ms G.723.130 ms
For example, to receive 20 ms of G.729 packets, set the parameter to 2.

CallFeatures

Disable/enable CallFeatures by setting each corresponding bit to 0 or 1. The lower 16 bits are for the Phone 1 port, and the upper 16 bits are for the Phone 2 port. Table 5-2 provides definitions of each bit.
The subscribed features that can be permanently disabled by the user are CLIP_CLIR, call waiting and Fax mode. A subscribed service enable/disabled by the user can be disabled/enabled dynamically on a per-call basis.

0xffffffff

Table 5-2 CallFeatures Parameter Bit Definitons

Bit Number 15

Definition Forward unconditionally Forward on busy Forward on no answer CLIP_CLIR Call waiting three-way calling Blind transfer Transfer with consultation. This service allows the user to transfer the remote party to a different number by first calling that number and consulting with the callee. Caller ID. This service enables the Cisco ATA 186 to generate a Caller ID signal to drive a Caller ID display device attached to the FXS line. Call return Message waiting indication Call Waiting Caller ID. This is available only if the Method bit in CallerIdMethod is set to Bellcore (FSK). Fax mode. This service allows the user to set the Cisco ATA to Fax mode on a per-call basis. For Fax mode, use the following settings:

List of Call-Progress Tone Parameters
The following list contains the names of the call-progress tone parameters:
DialTone BusyTone ReorderTone RingBackTone CallWaitTone AlertTone

Tone Parameter Syntax

Each tone is specified by nine intergers, as follows: ntone, freq0, freq1, level0, level1, steady, on-time, off-time, total-tone-time
ntone is the number of frequency components (0, 1 or 2). freq[0] (Hz) is the transformed frequency of the first frequency component (-32768 to 32767).
Only positive values can be configured to the Cisco ATA 186. For negative values, use the 16-bit 2s-complement value. For example, enter -1 as 65535 or 0xffff. freq[1] is the transformed frequency of the second frequency component (-32768 to 32767). level[0] is the transformed amplitude of the first frequency component (-32768 to 32767). level[1] is the transformed amplitude of the second frequency component (-32768 to 32767). steady controls whether the tone is constant or intermittent. A value of 1 indicates a steady tone and causes the Cisco ATA to ignore the on-time and off-time parameters. A value of 0 indicates an on/off tone pattern and causes the Cisco ATA to use the on-time and off-time parameters. on-time controls the length of time the tone is heard in milliseconds (ms) expressed as an integer from 0 to 0xffff sample at 8000 samples/second. off-time controls the length of time between audible tones in milliseconds (ms) expressed as an integer from 0 to 0xffff sample at 8000 samples/second. total-tone-time controls the length of time the tone is audible (0 to 0xffff). If this value is set to 0, the tone will play until another call event stops the tone. For DialTone, BusyTone, ReorderTone, and RingBackTone, the configurable value is the number of 10 ms (100 = 1 second) units. For the other tones, the value is the number of samples at 8000 samples/second, where the following information applies:
Frequency ranges from 0 to 4000 (Hz) Transformed Frequency = 32767 cos (2piFrequency/8000) Amplitude ranges from 0 to 32767 Transformed Amplitude = A 32767 sin (2piFrequency/8000)
The scaling factor A determines the volume level of the tone. To calculate scaling factors, see the How to Calculate Scaling Factors section on page 5-43.

Chapter 6

Call Commands Syntax

Context-Identifiers

Table 6-1 Context-Identifiers
Identifier A B C D E F G H I J K L M N O P Q R S T U V W X Y Z a b
Context (State of Cisco ATA) CONFERENCE PREDIAL PREDIAL_HOLDING CONNECTED CONNECTED_HOLDING CONNECTED_ALERTING HOLDING CONFIGURING CONFIGURING_HOLDING 3WAYCALLING CALLWAITING IDLE RINGING DIALING CALLING Reserved (ANSWERING) Reserved (CANCELING) Reserved (DISCONNECTING) WAITHOOK DIALING_HOLDING CALLING_HOLDING Reserved (ANSWERING_HOLDING) Reserved (HOLDING_HOLDING) Reserved (CANCELING_HOLDING) Reserved (DISCONNECTING_HOLDING) Reserved (HOLDING_ALERTING) WAITHOOK_ALERTING WAITHOOK_HOLDING
Input Sequence Identifiers
Table 6-2 Input Sequence Identifiers
Identifier 0-9,#* f o @ h S N D v
Input Sequence DTMF digits hook flash off-hook anytime; for example, @f means anytime hookflash occurs on-hook #|* 0|1|2|3|4|5|6|7|8|9 N|S a variable number (1 or more) of characters from the above list. It must be followed by a character which acts as the terminator of this variable part.

Action Identifiers

Table 6-3 Action Identifiers
Identifier A B C D E F G H I N O P Q R a b
Action NONE Seizure (User intends to dial or configure) Continue to dial Call Return Hold the active call Retrieve the waiting call Cancel the call attempt Disconnect the call Blind transfer the call to the number Go to configuration mode Release the call Answer the incoming call Transfer with consultation Say busy to the caller None Forward all calls to the given number
Call Commands Call Command Example

Table 6-3

Action Identifiers (continued)

c d e f g h i x y

Forward on busy to the given number Forward on no answer to the given number Cancel call forward CLIP for the next call CLIR for the next call Enable Call Waiting for the next call Disable Call Waiting for the next call Enable Fax Mode for the next call Disable Fax Mode for the next call

Call Command Example

In addition to call commands that you configure, the Cisco ATA has a default list of call commands to handle common call scenarios. Configured call commands overwrite default call commands. If any Context-Identifier or Input-Sequence elements appear in both the default Call Command string and the manually entered string, the manually entered value takes precedence. The following string shows a sample Call Command:
Bf;BAN;CA;CN;CAf;OF;Df;EB;I@f;OF;H@f;OA;Lo;BAf;BA;Mo;PA;ND;CAf;OA;Of;GA;Pf;HA;Qf;OA;Rf;OA; Sf;OA;TD;CAf;OF;Uf;GF;Vf;HF;Wf;FF;Xf;AF;Yf;AF;Zf;AP;bf;OF;af;OP;

In this section, the Call Command string is broken down into its components as follows:
Call Command Fragment; Context-Identifier Input-Sequence1; Action1 Action2; (optional) Input-Sequence2; Action1 Action2;
If you use a second input sequence, this sequence follows the Action Identifier pair without a separating semicolon.
Refer to the preceding tables to determine the meanings of the identifiers.
Example 6-3 Call Command String
Bf;BAN;CA; Predial hook-flash; Seizure NONE 0|1|.|9; Continue-to-dial NONE; CN;CAf;OF; Predial_Holding 0|1|.|9; Continue-to-dial NONE hook-flash; Release-the-call Retrieve-the-waiting-call; Df;EB; Connected hook-flash; Hold-the-active-call Seizure; I@f;OF; Configuring_Holding hook-flash (at any time); Release-the-call Retrieve-the-waiting-call; H@f;OA;
Chapter 6 Call Command Example
Configuring hook-flash (at any time); Release-the-call NONE; Lo;BAf;BA; Idle off-hook; Seizure NONE; hook-flash; Seizure NONE; Mo;PA; Ringing off-hook; Answer-the-incoming-call NONE; ND;CAf;OA Dialing 0|1|.|9|#|*; Continue-to-dial NONE hook-flash; Release-the-call NONE; Of;GA; Calling hook-flash; Cancel-the-call-attempt NONE; Pf;HA; Answering hook-flash; Disconnect-the-call NONE; Qf;OA; Canceling hook-flash; Release-the-call NONE; Rf;OA; Disconnecting hook-flash; Release-the-call NONE; Sf;OA; Waithook hook-flash; Release-the-call NONE; TD;CAf;OF; Dialing_Holding 0|1|.|9|#|*; Continue-to-dial NONE; hook-flash; Release-the-call NONE; Uf;GF; Calling_Holding hook-flash; Cancel-the-call-attempt Retrieve-the-waiting-call; Vf;HF; Answering_Holding hook-flash; Disconnect-the-call Retrieve-the-waiting-call; Wf;FF; Holding_Holding hook-flash; Retrieve-the-waiting-call Retrieve-the-waiting-call; Xf;AF; Canceling_Holding hook-flash; NONE Retrieve-the-waiting-call; Yf;AF; Disconnecting_Holding hook-flash; NONE Retrieve-the-waiting-call; Zf;AP; Holding_Alerting hook-flash; NONE Answering; bf;OF; Waithook_Holding hook-flash; Release-the-call Retrieve-the-waiting-call; af;OP; Waithook_Holding hook-flash; Release-the-call Answer-the-incoming-call;
Call Commands Call Command Behavior

Call Command Behavior

Table 6-4 summarizes differing Call Command behavior based on the U.S. and Sweden default call commands.
U.S. Call Command Default
Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67;gA*82;fA#90v#;OI;H#72v#;bA#74 v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;Uh;GQ;Af;AH;
Sweden Call Command Default

Perform the command: dial-peer voice tag voip Perform the command: modem passthrough {NSE [payload-type number] codec {g711law | g711alaw} [redundancy] | system}
The default of this command is: modem passthrough system When using the default configuration, the dial-peer fax pass-through configuration is defined by the voice service voip command. When the system option is used, no other parameters are available. When the NSE is configured in the fax pass-through command at the dial-peer level, the fax pass-through definition in the dial-peer command takes priority over the definition in the voice service voip command.
The payload-type number, codec, and redundancy parameters can also be used. For example, the command: modem passthrough NSE codec g711law means that the Cisco ATA will use the NSE payload-type number 100, G.711-law codec, and no redundancy in fax pass-through mode.
When setting up dial-peer for fax pass-through, it is necessary to set up a pair of dial-peers for inbound and outbound calls between the Cisco ATA and Cisco IOS gateways. You do this by specifying the destination-pattern and incoming-called number. The destination-pattern should point to the Cisco ATA, while the incoming-called number should apply to all numbers that the Cisco ATA is allowed to dial.
Disable Fax Relay Feature
Fax relay may be enabled by default for some IOS gateways. If you do not disable the fax relay feature, it may override the precedence of fax/modem pass-through and cause the fax transmission to fail. It is necessary to disable fax relay at the dial-peer or system level with the following command: fax rate disable

Chapter 7 Using FAX Mode

Using FAX Mode
Use fax mode when the gateways in the network do not support fax pass-through mode or dial-peer configuration. You can set one or both lines of the Cisco ATA to G.711-only fax mode. This mode allows the fax machine connected to the Cisco ATA to communicate directly with the far endpoint with no fax signaling event occurring between the two gateways. This section contains the following topics:
Configuring the Cisco ATA for Fax Mode, page 7-6 Configuring the Cisco ATA for Fax Mode on a Per-Call Basis, page 7-7 Configuring the Cisco IOS Gateway for Fax Mode, page 7-7
Configuring the Cisco ATA for Fax Mode
G.711-only fax mode operation requires configuration of one parameterAudioMode.
The AudioMode parameter is a 32-bit value. The lower 16 bits apply to the Phone 1 port of the Cisco ATA, and the upper 16 bits to the Phone 2 port. The following is an example of the Phone 1 port of the Cisco ATA configured for G.711-only fax mode:

0xXXXX0012

xxxx xxxx xxxx xxxx 0001 0010
Bit 0 = 0Disables G.711 silence suppression (VAD). Bit 1 = 1Uses G.711 only, does not user the low bit-rate codec. Bit 2 = 0Disables Fax CED tone detection. Bit 4 = 1, Bit 5 = 0DTMF transmission method: out-of-band through negotiation Bit 6 = Bit 7 = 0Hookflash transmission method: disables sending out hookflash

The values XXXX in the example do not apply to the Phone 1 port of the Cisco ATA. To configure the same value for the Phone 2 port of the Cisco ATA, the value would be 0x0012XXXX. The configuration of one port is independent from the configuration of the other port.
The AudioMode configuration overrides the values of the following three parameters: RxCodec, TxCodec, and LBRCodec. For example, if these three parameters are each set to 0 (for G.723), the Cisco ATA would still use G.711 if AudioMode is set to 0x00120012. With this configuration, the Cisco ATA sends both G.711-law and G.711A-law as preferred codecs to a peer voice gateway.
Configuring and Debugging Fax Services Debugging the Cisco ATA 186/188 Fax Services
Configuring the Cisco ATA for Fax Mode on a Per-Call Basis
The per-call-basis fax mode feature is only available for the H.323 and SIP protocols. If you want to activate fax mode on a per-call basis, configure the following parameters:
CallFeatures and PaidFeatures Bit 15 (for line1mask 0x8000) and Bit 31 (for line2mask 0x80000000) = 1: This sets the default to enable fax mode on a per-call basis. AudioMode Bit 2 = 0: This disables fax CED tone detection. CallCmd includes *99;xA (99 is the default; the value can be changed to any prefix code.)
To activate a call from your fax machine, enter *99 (default), then enter the telephone number to which you want to send the fax. The next call will automatically revert to normal mode.
Configuring the Cisco IOS Gateway for Fax Mode
On the Cisco gateway, disable both fax relay and fax pass-through at the dial-peer level or system level with the following commands:
Run the command: fax rate disable Run the command: no modem passthrough
Debugging the Cisco ATA 186/188 Fax Services
This section includes the following debugging topics for fax services:
Common Problems When Using IOS Gateways, page 7-7 Using prserv for Diagnosing Fax Problems, page 7-9 Using rtpcatch for Diagnosing Fax Problems, page 7-12
Common Problems When Using IOS Gateways
Table 7-1 lists typical problems and actions that might solve these problems for situations in which the Cisco ATA is using fax over a Cisco IOS gateway.
Chapter 7 Debugging the Cisco ATA 186/188 Fax Services

Table 7-1

Solving Common Fax Problems

Problem

The far-end gateway is not Cisco recommends IOS version 12.2 (11)T or higher for the Cisco 2600 loaded with correct and Cisco 3600, and IOS version 12.1 (3)T or higher for Cisco AS5300. software image. The Cisco 6608 supports both the NSE and NTE methods of fax pass-through mode, beginning with software version D004030145S16608. To use fax pass-through mode with the Cisco 6608, the user must select 6608 NSE mode, and the NSE payload type must be reconfigured to match the Cisco ATA. The Cisco ATA is not loaded with the proper software. Cisco recommends using software version 2.14 or higher.

<3.200> is the originating Cisco ATA, and <2.53> is the terminating gateway. Both sides initially use G.729. <2.53> gateway sends NTE signaling packets, then upspeeds to G.711-law. <3.200>The Cisco ATA switches to G.711-law also, but never sends NTE signaling packets. Fax transmission fails because <2.53> gateway does not receive any NTE packets, and it drops the fax call.
The Cisco ATA does not support the NTE signaling method and requires that the gateways use the NSE signaling method.

rtpcatch Limitations

rtpcatch performs optimally when analyzing capture files containing only one VoIP session. rtpcatch detects only G.711A, G.711-law, G.723, G.729, T.38, Cisco fax relay, modem pass-through with or without redundancy packets, RTCP packets and NSE packets.
rtpcatch can handle a maximum of 20 prserv ports using the -port option. rtpcatch may not detect T.38 packets correctly.
Upgrading the Cisco ATA Signaling Image
This section describes two methods for upgrading the Cisco ATA software for the SIP protocol:
Upgrading the Signaling Image from a TFTP Server, page 8-1This is the Cisco-recommended method for the SIP protocol. This method is the most efficient method and requires only a one-time configuration change. Upgrading the Signaling Image Manually, page 8-2This method can be used if you must manually upgrade the image of one Cisco ATA. However, this method is not the recommended upgrade method because it is not as simple as the TFTP method.
This section also describes procedures for verifying a successful image upgrade:
Confirming a Successful Signaling Image Upgrade, page 8-5Procedures for using your Web browser or the voice configuration menu are included.
Do not unplug the Cisco ATA while the function button is blinking. Doing so can cause permanent damage to the device. The function button blinks during an upgrade.
You can configure the Cisco ATA to automatically download the latest signaling image from the TFTP server. You do this configuring the parameter upgradecode in your Cisco ATA configuration file. (You also would use this procedure if you wanted to perform a cross-protocol signaling image upgrade.) For more information about setting up the configuration file, see the Creating Unique and Common Cisco ATA Configuration Files section on page 3-8.
Syntax of upgradecode Parameter
upgradecode:3,0x301,0x0400,0x0200,tftp_server_ip,69,image_id,image_file_name

Definitions

The hexadecimal values that precede the tftp_server_ip variable must always be the values shown in the syntax. tftp_server_ip is the TFTP server that contains the latest signaling image file.

A two-way voice path is established between Cisco ATA A and Cisco ATA B.

Step 11

Step 12 Step 13 Step 14 Step 15
100 TryingSIP server to Cisco ATA A BYESIP server to Cisco ATA B 200 OKCisco ATA B to SIP server 200 OKSIP server to Cisco ATA A

Table D-6 1.

INVITE sip:9000@192.168.2.97;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.3.175 From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone> Call-ID: 488337201@192.168.3.175 CSeq: 1 INVITE Contact: <sip:8000@192.168.3.175;user=phone;transport=udp> User-Agent: Cisco ATA v2.12 ata186 (0928a) Expires: 300 Content-Length: 253 Content-Type: application/sdp v=0 o=206154 IN IP4 192.168.3.175 s=ATA186 Call c=IN IP4 192.168.3.175 t=m=audio 10000 RTP/AVP a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SIP/2.Trying Via: SIP/2.0/UDP 192.168.3.175 Call-ID: 488337201@192.168.3.175 From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone> CSeq: 1 INVITE Content-Length: 0 INVITE sip:9000@192.168.2.194;user=phone SIP/2.0 Record-Route: <sip:9000@192.168.2.97:5060;user=phone;maddr=192.168.2.97> Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1 Via: SIP/2.0/UDP 192.168.3.175 From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone> Call-ID: 488337201@192.168.3.175 CSeq: 1 INVITE Contact: <sip:8000@192.168.3.175;user=phone;transport=udp> User-Agent: Cisco ATA v2.12 ata186 (0928a) Expires: 300 Content-Length: 253 Content-Type: application/sdp v=0 o=206154 IN IP4 192.168.3.175 s=ATA186 Call c=IN IP4 192.168.3.175 t=m=audio 10000 RTP/AVP a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

Table D-6 4.

Log Listings (continued)
SIP/2.Trying Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1 Via: SIP/2.0/UDP 192.168.3.175 Record-Route: <sip:9000@192.168.2.97:5060;user=phone;maddr=192.168.2.97> From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone>;tag=909616993 Call-ID: 488337201@192.168.3.175 CSeq: 1 INVITE Server: Cisco ATA v2.12 ata186 (0928a) Content-Length: 0 SIP/2.Ringing Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1 Via: SIP/2.0/UDP 192.168.3.175 Record-Route: <sip:9000@192.168.2.97:5060;user=phone;maddr=192.168.2.97> From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone>;tag=909616993 Call-ID: 488337201@192.168.3.175 CSeq: 1 INVITE Server: Cisco ATA v2.12 ata186 (0928a) Content-Length: 0 SIP/2.Ringing Via: SIP/2.0/UDP 192.168.3.175 Record-Route: <sip:9000@192.168.2.97:5060;user=phone;maddr=192.168.2.97> From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone>;tag=909616993 Call-ID: 488337201@192.168.3.175 CSeq: 1 INVITE Server: Cisco ATA v2.12 ata186 (0928a) Content-Length: 0 SIP/2.OK Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1 Via: SIP/2.0/UDP 192.168.3.175 Record-Route: <sip:9000@192.168.2.97:5060;user=phone;maddr=192.168.2.97 From: <sip:8000@192.168.2.97;user=phone>;tag=2819471139 To: <sip:9000@192.168.2.97;user=phone;tag=909616993 Call-ID: 488337201@192.168.3.175 CSeq: 1 INVITE Contact: <sip:9000@192.168.2.194;user=phone;transport=udp> Server: Cisco ATA v2.12 ata186 (0928a) Content-Length: 199 Content-Type: application/sdp v=0 o=206275 IN IP4 192.168.2.194 s=ATA186 Call c=IN IP4 192.168.2.194 t=m=audio 10000 RTP/AVP a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

NSE packets

NAT Server
Plain old telephone service. Basic telephone service supplying standard single-line telephones, telephone lines, and access to the PSTN. An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it. Public switched telephone network.

Proxy Server

Quality of Service. The capability of a network to provide better service to selected network traffic over various technologies, including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and 802.1 networks, SONET, and IP-routed networks that may use any or all of these underlying technologies. The primary goal of QoS is to provide priority including dedicated bandwidth, controlled jitter and latency (required by some real-time and interactive traffic), and improved loss characteristics.

Redirect Server

A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses, and returns these addresses to the client. It does not initiate its own SIP request nor accept calls. A registrar server is a server that accepts Register requests. A registrar is typically co-located with a proxy or redirect server and may offer location services. Network layer device that uses one or more metrics to determine the optimal path along which network traffic should be forwarded. Routers forward packets from one network to another based on network layer information. Occasionally called a gateway (although this definition of gateway is becoming increasingly outdated). Compare with gateway. Real-Time Transport Protocol. One of the IPv6 protocols. RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides services such as payload type identification, sequence numbering, timestamping, and delivery monitoring to real-time applications.

Registrar Server

router

SCCP SDP

3-15 5-7 5-28 3-2 3-3

trace-features configuration TraceFlags parameter
transfer with consultation
statically configuring various IP addresses StaticIP parameter StaticNetMask

3-15 5-8

transfer-with-consultation configuration setting transmitting-audio codec settings troubleshooting general tips installation
5-34 9-1 9-3 9-3 5-28 5-22
StaticNetMask parameter StaticRoute

3-15, 4-7 5-7

static network-router-probing configuration setting StaticRoute parameter stuttering dial tone

upgrade issues

two-way cut-through configuration setting TxCodec parameter

5-8 5-22

subnet mask static configuration supplementary services cancelling common

A-1 A-1

UDPTOS parameter UID0 parameter UID1 parameter
5-9 5-9 5-3, 9-5 5-35 5-32

UIPassword parameter

UIPassword promt configuration setting
unconditional call forwarding unique configuration file upgrading software using executable file
circuit breaker (15A) installation

8-1 2-2 2-2

3-9, 3-12
upgrading firmware from TFTP server
lightning activity No. 26 AWG
main disconnecting device
upgrading software from TFTP server upgrading the signaling image UseLoginID parameter User agent client (UAC) User agent server (UAS) user configurable timeout User ID
5-9 5-3 5-11 1-3 1-3 4-10 3-22

product disposal

2-2 5-34 4-6
web configuration disabling Web interface access disabled
User Interface (UI) Parameters UseSIP parameter UseTFTP parameter

5-14 5-4

VIA header
VLAN CoS bit value (802.1P priority) for TCP packets 5-35 VLAN CoS bit value (802.1P priority) for UDP packets 5-35 VLAN ID

5-35 3-4 5-34

VLAN ID example
VLAN IP encapsulation setting VLAN-related parameters VLAN Setting parameter VLAN tagging

3-2 3-16 3-3 5-35

voice configuration menu Voice Menu Codes configuration parameters information options software upgrade

B-1 B-4

WAN address of attached router/NAT warnings