Cisco 3600 ios download

Each feature set contains a specific set of Cisco IOS features. Cisco IOS Release Note If you have a Cisco. These tables all use the following conventions:. For example, 2 XA2 means a feature was introduced in The following sections list the new hardware and software features supported by the Cisco series for Release Before the call admission control feature, gateways did not have a mechanism to gracefully prevent calls from entering when certain resources were not available to process the call.

This causes the new call to fail with unreported behavior, and could potentially cause the calls that are in progress to have quality related problems.

This feature set provides the ability to support resource-based call admission control processes. These resources include system resources such as CPU, memory, and call volume, and interface resources such as call volume. If system resources are not available to admit the call, two kinds of actions are provided: system denial which busyouts all of T1 or E1 or per call denial which disconnects, hairpins, or plays a message or tone.

If the interface-based resource is not available to admit the call, the call is dropped from the session protocol such as H. This feature allows a user to configure call admission thresholds for local resources as well as memory and CPU resources.

The list of local resources that are configured for call admission are described in the command description of "call threshold poll-interval. With the call admission command, a user is allowed to configure two thresholds, high and low, for each resource.

Call treatment is triggered when the current value of a resource goes beyond the configured high. The call treatment remains in effect until current resource value falls below the configured low. Having high and low thresholds prevents call admission flapping and provides hysteresis in call admission decision making. With the call spike command, a user is allowed to configure the limit for incoming calls during a specified time period.

A call spike is the term for when a large number of incoming calls arrive from the PSTN in a very short period of time for example incoming calls in 10 milliseconds.

With the call treatment command, users are allowed to select how the call should be treated when local resources are not available to handle the call. For example, when the current resource value for any one of the configured triggers for call admission has reached beyond the configured threshold, the call treatment choices are as follows:.

This feature set supports the autobusyout feature where channels are busied out when local resources are not available to handle the call. The user defines the congestion thresholds based on the configured network. This functionality enables the service provider to give a reasonable guarantee about the quality of the conversation to their VoIP users at the time of call admission. Based on the packet loss, delay, and jitter encountered by these probes, an ICPIF or delay and loss values are calculated.

Only G. Calls of all other codecs are emulated by a G. The Cisco H. The enhancements in this release include:. The LCF responses indicate an alternate route to that endpoint. The GK determines the best route to an endpoint based on which route has the lowest cost and the highest priority.

I have this problem too. All forum topics Previous Topic Next Topic. Marvin Rhoads. VIP Community Legend. If that's an old , those have not been sold by Cisco since In response to Marvin Rhoads. Will any of the newer software work on it or is there open source software for it? Richard Burts. Hall of Fame Guru. In response to kevpfeifle1.

In response to Richard Burts. It does not see the bank of fe ports. Please share output of the following commands: show version show inventory show ip interface brief.

Show inventory does not work and I only plugged in one power supply. It's running version It is not being recognized by the router IOS. Ringback is not available for calls coming in from a PSTN. IP phones do not ringback on the network side. Alignment traceback when transfer and conference call. Tracebacks are observed during conference and at times, during regular calls. The tracebacks are on misaligned accesses to memory.

They are subsequently corrected by the hardware but prints to console are annoying and can be viewed as serious potential issues. Bad voice-quality during conference call. Conferencing a-law and u-law mixes doesn't work as some legs in ITS end up with a-law and some with u-law which gives bad voice quality. This fix is only applicable for a network that has a mixture of A-law and U-law codecs configured in the network.

Configure all gateways in the network to use same codec type of either A-law or U-law. This problem is only on Cisco series platform.

The hardware version, revision information, and other conditions are displayed wrongly. The port does not go on-hook when a supervisory disconnect tone is sent from the PSTN.

If the caller hangs up at this point, the PSTN sends a supervisory disconnect tone to the FXO where no action is taken which results in not freeing up the port immediately.

The IP phone locks during a call to a call forward busy number. When a call is made from phone A to another phone phone C , the call can be answered normally. When a second call is placed from another phone to phone A via a loopback-directory number loopback-dn , the call is forwarded to phone B. As this call is received, the original call is incorrectly cleared by the node on phone C while the call is still shown as "up" on phone A.

When this condition occurs, phone A is no longer able to place or receive calls and has to be powered down to be restored to working condition. This condition affects only the first IP phone in the internal control table on the router. To prevent this condition from occurring when the router is operating in the ITS mode, add a dummy phone entry as the first IP phone or "ephone 1" in the internal control table on the router to prevent an active phone from being listed as the first IP phone in the internal control table of the router.

Reload the router to ensure that the dummy phone occupies the first position in the control table. There is currently no workaround to this condition if the router is operating in the Survivable Remote Site Telephony SRST mode call manager fall back.

When using loopback-dns for outgoing calls, the router presents a ringing tone to the IP phone caller as soon as the call has been routed and before the state of the called number is known. If the called number is busy, this can result in a ringing and then busy tone being played to the caller. This does not happen if loopback-dns are not used, or if they are used for incoming calls.

When a Keyswitch IP phone user receives an incoming call, and they then attempt to transfer that call, but while dialling they realize they are dialling a wrong number, the transferrer has two options. If they incorrectly select "New Call", the phone displays the previous dialed digits. If the caller then dials more digits, these appear after the previous digits. The phone however only dials the digits presented after the "New Call" button was pressed, and the call is successful.

The problem is observed only when the second call clears, and only if another IP phone is used in addition to the IP phone which set the conference up. Voice path is re-established if the phone that initiated the conference temporarily places the remaining caller on-hold and then immediately resumes the call. This release integrates the DSPware 3. Upgrading to the DSPware 3. This section describes only severity 1 and 2 caveats and select severity 3 caveats.

An error can occur with management protocol processing. Please use the following URL for further information:. Problem: Lane client does not become operational. First Lane packet is dropped because of the CRC check failed.

All the AAL5 connections are going to have the same problem of losing first packet. The Cisco and Cisco routers return the wrong power supply type when using with the ciscoEnvMonSupplySource snmp query. This condition is noticeable only on low end systems that support multiple banksizes for internal flash. Problem: The configured as a voice gateway will not register to it's gatekeeper after a reload. This can be an intermittent problem, and may not occur on all 's.

Symptom: If you do a debug ras on the gateway there are no ras messages being sent to the gatekeeper. Workaround: If you do a shut, no shut on the fastethernet interface, the gateway will register to the gatekeeper.

Problem is occurring when the customer is attempting to receive dial-tone on ISDN trunks. The method that the customer uses to obtain this dial-tone is to configure the ISDN trunks for overlap-receiving, and send in a setup message with no called party number.

This causes the dial-tone from the router to not be heard. A new keyword has been added to the max-dn command allows you to set IP phones to dual-line mode. Each dual-line IP phone must have one voice port and two channels to handle two independent calls.

This mode enables call waiting, call transfer, and conference functions on a single ephone-dn. Dual-line mode works with all phone types.

The date format on Cisco IP phone displays can be configured with the following two additional formats:. A ringing timeout default can be configured for extensions on which no-answer call forwarding has not been enabled. Expiration of the timeout causes incoming calls to return a disconnect code to the caller.

This mechanism provides protection against hung calls for inbound calls received over interfaces such as foreign exchange office FXO that do not have forward-disconnect supervision. The show ephone command has been enhanced to display the following:. Diagnostic messages are added to the system log whenever a phone registers or unregisters from Cisco SRST. For conferencing to be available, an IP phone must have a minimum of two lines connected to one or more buttons. Several international languages and call-progress tone sets are newly supported.

The set of supported languages varies by phone type. Cisco CME Version 3. The auto assign command specifies a range of extension numbers to which newly discovered IP phones are automatically assigned. This method is useful when you have a phone setup in which each phone is assigned a separate, unique extension number.

Call pickup allows phone users to retrieve calls from other extension numbers by using the PickUp soft key and dialing the ringing number. When extensions are assigned to pickup groups, other members of the group can retrieve incoming calls using fewer keystrokes. When night service is active, incoming calls to designated night-service extension numbers will also ring on other phones that are designated as night-service phones. Phone users at the other phones can use call pickup to retrieve the incoming calls.

Call blocking to prevent the unauthorized use of phones is implemented by matching calls to a specified digit pattern during a specified time period. Up to 32 patterns of digits can be specified. Individual phones can be exempted from call blocking, and individual user logins can override call blocking if they are configured. Ephone hunt groups provide the ability to direct incoming calls for a specific number the ephone hunt group pilot number to a defined group of extensions. Incoming calls are redirected on busy or no answer from extension to extension in the list until they are answered or they reach the number that was defined as the final number.

Secondary dial tone is generated when a phone user dials a predefined digit. The tone terminates when additional digits are dialed. For example, you can configure a secondary dial tone to be heard after the number 9 is dialed to reach an external line. Phone users access the list of local speed-dial numbers from the Directories button. Phone users access their list of personal speed-dial numbers from the Directories button. Cisco IP Phone and Cisco IP Phone users can enter account codes during call setup or while connected to an active call, using the Acct soft key.

Account codes are inserted into call detail records CDRs on the CME router for later interpretation by billing software. This feature allows callers who dial a busy extension number to request a callback from the system when a called number that was busy is free.

Callers can also request callbacks for extensions that do not answer and the system will notify them after the called phone is next used. When DND is enabled, incoming calls do not ring on the phone, but do provide visual alerting and call information and can be answered if desired.

A display message indicates that DND is in effect. Call forwarding on busy and no answer operates the same as without DND. Certain PSTN services, such as three-way calling and call waiting, require hookflash intervention from a phone user. The Flash soft key is enabled using the fxo hook-flash command. Dual-line extensions are available to handle call-waiting, call transfer, or conferencing using a single button.

An extension ephone-dn overlay allows more than one ephone-dn to use the same physical line button on an IP phone.

Overlaid ephone-dns can be used to receive incoming calls and place outgoing calls. In particular, the GUI facilitates the routine adds and changes associated with employee turnover, allowing these changes to be performed by non-technical staff.

This person does not have to be trained in Cisco IOS software. The label support feature allows you to enter a meaningful text string to view in the display adjacent to an extension button on an IP phone rather than the extension number that is associated with that button.

For multi-button phones and expansion modules, the buttons for extensions that are shared with other phones can be designated as monitor buttons, which show the status of those extensions on the other phones. When not in use, a monitor line can be used with the Transfer soft key to quickly transfer a call. The Cisco CME system automatically creates a local phone directory based on the telephone numbers that are assigned during the configuration of extensions and phones.

Additional entries to the local CME directory can be made using the directory entry command. The silent ring feature allows you to designate phone buttons that do not emit an audible ring when they receive incoming calls. Although this feature is supported by all phone types, it is most useful on phone buttons that are used to display the activity of shared lines, which are typically found on the Cisco IP Phone and Cisco IP Phone Expansion Module Approximately 35 new and modified commands are described in the Command Reference at:.

This feature implements the downloading of region-specific tones and the associated frequencies, amplitudes, and cadences using XML-based configuration files during gateway registration. The feature supports dual tones and sequential tones.

Cisco CallManager performs signal and call processing. When Cisco CallManager requests a specific tone, the gateway references the custom tone table associated with the network locale of the voice port. When the gateway registers to Cisco CallManager, or if the gateway restarts or resets, the network locale for each port is downloaded to the gateway. Once the custom tone specification is downloaded to the gateway, it can also be used in H.

Only one gateway supports the download of up to two custom tones, that is, no more than two custom tone tables will be downloaded to one gateway even if there are more that two countries or regions configured for the gateway. The G. All other platforms continue to use the Cisco-proprietary ms EC by default. The feature support on the and port Ethernet switch network modules has been significantly enhanced in Cisco IOS Release Supports new standard. IEEE BackboneFast provides fast convergence in the network backbone after a spanning-tree topology change occurs.

Internet Group Management Protocol IGMP snooping constrains the flooding of multicast traffic by dynamically configuring the interfaces so that multicast traffic is forwarded only to those interfaces associated with IP multicast devices. Per-port storm control prevents broadcast, multicast, and unicast storms. Per-port storm-control uses rising and falling thresholds to block and then restore the forwarding of broadcast, unicast, or multicast packets. You can also set the switch to shut down the port when the rising threshold is reached.

A routed port is a physical port that acts like a port on a router; it does not have to be connected to a router. A routed port is not associated with a particular VLAN, as is an access port. A routed port behaves like a regular router interface, except that it does not support subinterfaces.

Routed ports can be configured with a Layer 3 routing protocol. Fallback bridging forwards traffic that the multilayer switch does not route and forwards traffic belonging to a nonroutable protocol such as DECnet. There are 47 new Cisco IOS commands that support the feature enhancements. For additional information on the feature enhancements, also refer to the and Port Ethernet Switch Module for Cisco Series, Cisco Series, and Cisco Series feature module at:.

Should connection to the primary call manager fail, call processing reverts to a backup call manager until the connection to the primary is restored. Should connections to the primary and all backups fail, call processing reverts to H. When a connection is restored, call processing reverts to the primary or other available call manager and to MGCP. You need a supported Cisco series, router equipped with the following:. You need one or more Cisco CallManager systems, Version 3.

This feature delivers Private Line Automatic Ringdown for the connection of turrets for the financial industry—primarily for corporations and enterprises that use turrets and POTS telephones for trading. Implementation of this feature ensures that a call between traders on a PLAR connection will be maintained if one of the traders goes on-hook or on-hold.

This new capability also ensures that bandwidth is used only when needed. For additional command syntax and configuration information, refer to the Private Line Automatic Ringdown for Trading Turrets feature module at:. When a voice port is configured with an incorrect destination number that may or may not be a valid number, the call may not perform as expected. There is no cross-checking for turret PLAR from the origination voice port, but there is a check on the terminating voice port to prevent accepting a call from a calling party that is not preconfigured.

These network modules provide the ability to directly connect the PSTN and legacy telephony equipment to Cisco XM series, Cisco series, and Cisco series modular access routers, enabling important applications such as IP telephony, toll bypass, and full gateway integration.

These network modules support the following interface cards:. Features supported in this release include the following:. In addition to continuing support for configuring a fixed number of channels per DSP, the flex option enables the DSP to handle a flexible number of channels.

The total number of supported channels varies from 6 to 16, depending on which codec is used for a call. All the signaling is transparently sent between the analog voice port and DS0 time slot, and will not be seen by the higher layer voice software. The following Cisco IOS commands are introduced or modified to support this feature:. Cisco IOS software images are subject to deferral.

Cisco recommends that you view the deferral notices at the following location to determine if your software release is affected:.