CUBE H323 to H323 Advice

[e.parsonage]

Hi There,

I wonder if someone could give me some advise on the best way to configure a CUBE which is currently not working as expected, I have a CUCM cluster v5.x and a pair of CUBEs (configured as H323 gateways) connected via H323 (non gatekeeper) trunk to a service provider.

The problem i have is that on outbound calls I can setup a call and can see Tx and Rx packet counts going up on the ip phones but have no audio.........

I believe the problem is to do with MTP and I will need to configure MTP resources locally on the CUBE and terminate the 2 legs of the calls on the CUBE as the service provider's prefered codec is G729 I don't believe local (CUBE) transcoding is required.

My question is firstly has anyone done anything like this and is willing to share configuration, and secondly does this seem to be the correct route of investigation or am I missing something and should be concenrating else where?

Thanks in advance

Ellis

[Chris Deren]

You should not be needing any MTPs for this type of setup. Can you elaborate on your topology and what type of circuit is the telco providing? Is it a SIP trunk?  Posting configuration would be helpful. Is your Region between the phone and CUBE configured to use G729?

How is routing performed, are you using NATing anywhere?

Chris

[Maxim Denisov]

Hi Ellis,

Check that RTP addresses are correct on both legs in show call history voice command output. Also check the rx and tx bytes count on both legs if media flow-through is configured.

MTP is not needed. Below is my config for calls between H.323 device and SIP

SBC2#sh run | sec voice service voip

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

supplementary-service media-renegotiate

fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw

h323

  no call service stop

  bearercap-ie signaling-only standard

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  referto-passing

  source filter

  midcall-signaling passthru

  g729 annexb-all

  no call service stop

SBC2#

dial-peer voice 2002 voip

permission orig

description #-- Incoming from Infinity

translation-profile incoming DP-FROM-INFINITY

huntstop

voice-class codec 1

incoming called-number ^DACCBA#.T

dtmf-relay rtp-nte digit-drop

dtmf-interworking rtp-nte

fax protocol pass-through g711alaw

no vad

dial-peer voice 2001 voip

permission orig

description #-- Incoming from GLDN

translation-profile incoming DP-FROM-GLDN

huntstop

voice-class codec 1

incoming called-number ^AACBAD#.T

dtmf-relay rtp-nte digit-drop

dtmf-interworking rtp-nte

fax protocol pass-through g711alaw

no vad

dial-peer voice 201 voip

permission term

description #-- UA numbers pool 1

huntstop

preference 2

destination-pattern +3804422214[4,5][0-9]$

voice-class codec 1

session target ipv4:10.38.10.10

dtmf-relay h245-signal

dial-peer voice 38 voip

permission term

translation-profile outgoing TO-GLDN

huntstop

destination-pattern +38..........$

voice-class codec 38

session protocol sipv2

session target ipv4:89.162.254.10

dtmf-relay rtp-nte

fax protocol pass-through g711alaw

no vad

Regards,

Maxim

[Marwan ALshawi]

did you check the routing from the Gateway to the phones if it works in both directions ?