sip call overflow

Unanswered Question
Jul 6th, 2008
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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

!


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Overall Rating: 4 (1 ratings)
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paolo bevilacqua Sun, 07/06/2008 - 09:08
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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.



anitachoi3 Mon, 07/07/2008 - 07:39
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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




paolo bevilacqua Mon, 07/07/2008 - 09:37
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Yes, most DP settings work the same no matter what the protocol used.


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

anitachoi3 Thu, 07/10/2008 - 09:00
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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

paolo bevilacqua Thu, 07/10/2008 - 11:15
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Hi,

SIP timers and counters are configured under sip-ua.



anitachoi3 Fri, 07/11/2008 - 17:10
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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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