Alternative Routing Parameters

'3xx Use Alt Route Reasons'

configure voip > sip-definition settings > 3xx-use-alt-route

[UseAltRouteReasonsFor3xx]

Defines the handling of received SIP 3xx responses regarding call redirection to listed contacts in the Contact header.

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[0] No = (Default) Upon receipt of a 3xx response, the device tries each contact, one by one, listed in the Contact headers, until a successful destination is found. However, if a contact responds with a 486 or 600, the device doesn't try to redirect the call to next contact, and drops the call.

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[1] No if 6xx = Upon receipt of a 3xx response, the device tries each contact, one by one, listed in the Contact headers. However, if a 6xx Global Failure response is received during this process (e.g., 600 Busy Everywhere) the device doesn't try to redirect the call to the next contact, and drops the call.

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[2] Yes = Upon receipt of a 3xx response, the device redirects the call to the first contact listed in the Contact header. If the contact responds with a SIP response that is defined in the Reasons for Tel-to-IP Alternative Routing table, the device tries to redirect the call to the next contact, and so on. If a contact responds with a response that is not configured in the table, the device doesn't try to redirect the call to the next contact, and drops the call.

'Redundant Routing Mode'

configure voip > gateway routing settings > redundant-routing-m

[RedundantRoutingMode]

Determines the type of redundant routing mechanism when a call can’t be completed using the main route.

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[0] Disable = No redundant routing is used. If the call can’t be completed using the main route (using the active Proxy or the first matching rule in the Routing table), the call is disconnected.

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[1] Routing Table = (Default) Routing table is used to locate a redundant route.

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[2] Proxy = Proxy list is used to locate a redundant route.

Note: To implement the Redundant Routing Mode mechanism, you first need to configure the parameter [AltRouteCauseTEL2IP] (Reasons for Alternative Routing table), as described in Alternative Tel-to-IP Routing Based on SIP Responses.

configure voip > gateway manipulation settings > alt-map-tel-to-ip

[EnableAltMapTel2IP]

Enables different Tel-to-IP destination number manipulation rules per routing rule when several (up to three) Tel-to-IP routing rules are defined and if alternative routing using release causes is used. For example, if an INVITE message for a Tel-to-IP call is returned with a SIP 404 Not Found response, the call can be re-sent to a different destination number, configured by the [NumberMapTel2IP] parameter.

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[0] = Disable (default)

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[1] = Enable

[TR104FXOSwitchover]

Enables the device to automatically switch the destination of an FXS call from the FXO (PSTN) to the IP (SIP Trunk) when the PSTN disconnects the FXS subscriber.

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[0] = (Default) Disable

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[1] = Enable

For more information, see Alternative Routing from FXO to IP.

Note: The parameter is applicable only to analog interfaces.

'Alternative Routing Tone Duration'

configure voip > gateway routing settings > alt-rte-tone-duration

[AltRoutingToneDuration]

Defines the duration (in milliseconds) for which the device plays a tone to the endpoint on each attempt for Tel-to-IP alternative routing. When the device finishes playing the tone, a new SIP INVITE message is sent to the new IP destination. The tone played is the call forward tone (Tone Type #25 in the CPT file).

The valid range is 0 to 20,000. The default is 0 (i.e., no tone is played).

Note:

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The parameter is applicable only to analog interfaces.

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The parameter is applicable only to Tel-to-IP alternative routing based on SIP responses (see Alternative Routing Based on SIP Responses).