Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. And see here for current known issues.

New Member

sip call over flow

Hi,

I configure the as5400 with two dial peers (1st and 2nd) for provider A and two dial peers (3rd and 4th) for provider B. Both of them offer sip calls.

If 1st dial peer for provider A is not working, I would like the call flow to 2nd dial per to provider A.

If 3rd dial peer for provider B is not working, I would like the call flow to 4th dial per to provider B.

How do I configure it? any sample configuration?

rdgs

7 REPLIES
New Member

Re: sip call over flow

set the prefernce on the dialpeer

New Member

Re: sip call over flow

Hi,

I add it but the call will not flow to second dial-peer when first dial-peer not working. Below is my config

R1

dial-peer voice 6501 voip

description provider A

tone ringback alert-no-PI

service session

destination-pattern 65.

no huntstop

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.

pref 1

no huntstop

max-conn 108

session protocol sipv2

session target ipv4:205.x.x.x

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

!

dial-peer voice 4401 voip

description provider B

tone ringback alert-no-PI

service session

destination-pattern 44.

no huntstop

max-conn 108

session protocol sipv2

session target ipv4:192.168.24.32

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

!

dial-peer voice 4402 voip

description provider B

tone ringback alert-no-PI

service session

pref 1

destination-pattern 44.

no huntstop

session protocol sipv2

session target ipv4:134.x.x.x

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

If I shutdown the 1st and 3rd dial-peer

, the call can flow to next dial-peer. How do I config the overflow automatically?

rdgs

New Member

Re: sip call over flow

ok, you have to specify the hunt by putting this command: dial-peer hunt ( 0-7)

New Member

Re: sip call over flow

Hi,

Sorry that I do not have any idea regarding "dial-peer hunt ( 0-7)"? what "0-7" refer to? please advise

rdgs

New Member

Re: sip call over flow

the 0-7 mean the following:

0 - Longest match in phone number, explicit preference, random selection.

1 - Longest match in phone number, explicit preference, least recent use.

2 - Explicit preference, longest match in phone number, random selection.

3 - Explicit preference, longest match in phone number, least recent use.

4 - Least recent use, longest match in phone number, explicit preference.

5 - Least recent use, explicit preference, longest match in phone number.

6 - Random selection.

7 - Least recent use.

New Member

Re: sip call over flow

Hi,

I should add below command:

R1

!

dial-peer hunt 2

!

I would like the call hunt to next dial-peer "only if" the first dial-peer is down (not working). Other conditon e.g. busy, no answer, wrong number.... , it keeps on going to the first dial-peer.

base on above requirment, any missing?

Thanks

rdgs

New Member

Re: sip call over flow

Hi,

I configure

dial-peer hunt 2

to router and find that all calls get busy if the first dial-peer is shutdown.

anything I missing?

rdgs

196
Views
0
Helpful
7
Replies
CreatePlease login to create content