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New Member

H323 VoIP peer not going to second preference.

I've got a SIP trunk from the carrier coming into a VGW/CUBE. The CUBE converts to H323 to CM 4.1(3). So I have my two VoIP dial-peers for each CM for protocol H323 for incoming calls back to CM. I want to verify that when my first CM goes down, the backup VoIP peer is selected and will route to the backup CM. So I built an access list denying IP any to the primary CM and placed on the CUBE H323 GW. Now the SIP call setup from the carrier keeps re-initializing and the outbound call leg (to CM) never appears to establish amd the phone never rings. debug voice dialpeer shows matching on both dialpeers. I've got my h323 voice class set up with timers. I tried explicitly setting the preference and changing the dialpeer hunt option to use preference first. Any ideas?

THANKS!

!

!

voice class h323 1

h225 timeout tcp establish 1

h225 timeout connect 60

h225 timeout setup 1

!

!

!

dial-peer voice 1 voip

preference 1

destination-pattern .T

voice-class h323 1

session target ipv4:10.1.1.1

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 2 voip

preference 1

destination-pattern .T

session protocol sipv2

session target ipv4:192.168.1.1

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3 voip

preference 2

destination-pattern .T

voice-class h323 1

session target ipv4:10.1.1.2

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

!

2 REPLIES
New Member

Re: H323 VoIP peer not going to second preference.

I would think you need to be more specific in your destination pattern.

If you dial 9 to call out on your Call Manager I would send that 9 to the CUBE and then put something like this in.

voice translation-rule 1

rule 1 /^9/ //

voice translation-profile strip9

translate called 1

dial-peer voice 200 voip

description Incoming From Call Manager Voice

voice-class codec 1

incoming called-number 9T

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 201 voip

description voip dial peer Out to SIP trunk

translation-profile outgoing strip9

destination-pattern 9T

voice-class codec 1

session protocol sipv2

session target ipv4:192.168.1.1

dtmf-relay rtp-nte digit-drop

no vad

And if your DID range where something like 12125553100 - 12125553199 I would put something like this

dial-peer voice 100 voip

description In from SIP voice

voice-class codec 1

voice-class h323 1

incoming called-number 12125553...

dtmf-relay h245-alphanumeric rtp-nte

no vad

!

dial-peer voice 101 voip

preference 1

description out to Call Manager voice

destination-pattern 12125553...

voice-class codec 1

voice-class h323 1

session target ipv4:10.1.1.1

dtmf-relay h245-alphanumeric

no vad

dial-peer voice 102 voip

preference 1

description out to Call Manager voice

destination-pattern 12125553...

voice-class codec 1

voice-class h323 1

session target ipv4:10.1.1.2

dtmf-relay h245-alphanumeric

no vad

Hope that helps

Re: H323 VoIP peer not going to second preference.

This isn't a good configuration.

You need to understand that CUBE does not take protocols in to account when looking for dial peers - it is very simplistic.

It will only go by preference and destination-pattern here. In this case, your SIP trunk on dial peer 2 and your CCM on dial peer 1 are equal-cost dial peers. In this scenario, it is possible to load balance between these two dial peers.

You should have the DID range for your CCM in your CCM dial peers. If it is not contiguous, you can use multiple dial peers.

When you do this, only those two dial peers will be matched and you will have fail-over from dial peer 1 to 3.

hth,

nick

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