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Helpful
6
Replies

sip call overflow

anitachoi3
Level 1
Level 1

Hi,

I have one AS5400XM, it handles the sip calls and routes the call to providers. The first route is provider A, second is provider B, the last is provider C. I find that the call cannot overflow to provider B and C. any missing?

rgds

The dial-peer setting is

R1

dial-peer voice 6501 voip

description provider A

tone ringback alert-no-PI

service session

destination-pattern 65.

max-conn 108

session protocol sipv2

session target ipv4:192.168.12.32

dtmf-relay rtp-nte h245-signal h245-alphanumeric

!

dial-peer voice 6502 voip

description provider A

tone ringback alert-no-PI

service session

destination-pattern 65.

max-conn 108

session protocol sipv2

session target ipv4:192.168.12.33

dtmf-relay rtp-nte h245-signal h245-alphanumeric

!

dial-peer voice 6503 voip

description provider B

tone ringback alert-no-PI

service session

pref 1

destination-pattern 65.

max-conn 108

session protocol sipv2

session target ipv4:192.168.24.32

dtmf-relay rtp-nte h245-signal h245-alphanumeric

!

dial-peer voice 6504 voip

description provider B

tone ringback alert-no-PI

service session

pref 1

destination-pattern 65.

session protocol sipv2

session target ipv4:192.168.24.33

dtmf-relay rtp-nte h245-signal h245-alphanumeric

!

dial-peer voice 6505 voip

description provider C

tone ringback alert-no-PI

service session

pref 2

destination-pattern 65.

max-conn 108

session protocol sipv2

session target ipv4:192.168.36.32

dtmf-relay rtp-nte h245-signal h245-alphanumeric

!

dial-peer voice 6506 voip

description provider C

tone ringback alert-no-PI

service session

pref 2

destination-pattern 65.

max-conn 108

session protocol sipv2

session target ipv4:192.168.36.32

dtmf-relay rtp-nte h245-signal h245-alphanumeric

!

6 Replies 6

paolo bevilacqua
Hall of Fame
Hall of Fame

Hi, you need "no huntstop" configured for all the DPs.

Also note DP 6504 has no max-conn configured so it won't stop accepting calls there.

Hi,

The correct setting should be with "no huntstop". this setting can work in sip call. Does it support h323 call?

rdgs

R1

dial-peer voice 6501 voip

no huntstop

dial-peer voice 6502 voip

no huntstop

dial-peer voice 6503 voip

no huntstop

dial-peer voice 6504 voip

no huntstop

max-conn 108

dial-peer voice 6505 voip

no huntstop

dial-peer voice 6506 voip

no huntstop

Yes, most DP settings work the same no matter what the protocol used.

As an appreciation to those providing answers, please rate useful posts!

Hi,

If the proivder A sip servers down, does it overflow to the providers B? if so, which timer to control the overflow when first Dial-peer down?

e.g.

1 second, the call will overflow,

or 3 seconds later, the call will overflow

rdgs

Hi,

SIP timers and counters are configured under sip-ua.

Hi,

By default setting, the proivder A goes down, does the sip call flow to provider B base on the config?

rdgs

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