Config H323 between CUCM and GW 2900

[Pablo Escribano Aparicio]

I am configuring H323 protocol between CUCM and other PBX for voice service. I have managed to make some remote sites work, but some others do not work. The flow would be the following:  CUCM --> Main GW --> Remote GW  --> Remote PBX, all through H323 protocol. 

IOS  GW 2900: c2900-universalk9-mz.SPA.153-3.M5

********************************************
Config H323:

!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice rtp send-recv
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  no h225 timeout keepalive
  h225 connect-passthru
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729br8
 codec preference 4 g729r8
!
voice class h323 1
  h225 timeout tcp establish 2
  h225 timeout setup 2
  call start slow
  telephony-service ccm-compatible
  ccm-compatible

**********************************************

Config  MTP y TRANSCODE (registered and working):

!
sccp ccm group 1
 bind interface GigabitEthernet0/0
 associate ccm 1 priority 1
 associate profile 101 register GW_XXXX
 associate profile 100 register GW_XXXY
 keepalive retries 5
 switchover method immediate
 switchback method immediate
 switchback interval 15
!
dspfarm profile 100 transcode
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec g729r8
 codec pass-through
 codec g729br8
 maximum sessions 15
 associate application SCCP
!
dspfarm profile 101 mtp
 codec g711ulaw
 maximum sessions hardware 29
 associate application SCCP

**********************************************

Config Dial Peer example fail:

dial-peer voice 440 voip
 description **** RICO ****
 max-conn 8
 destination-pattern 17T
 translate-outgoing called 210
 session target ipv4:XXX.XXX.XXX.XXX
 incoming called-number 3491.......
 voice-class codec 1
 voice-class h323 1

**********************************************

Following is the example of one connection with problem. The problem is that when we call we can hear three tones and then the busy tone. The correct behavior is that we shoudl hear the locution at first tone.

Attached Debugs:

debug ccsip mess
debug voip ccapi inout

If you need any different debug, please let me know.

thanks and regards¡¡

[Chris Deren]

Call disconnects with reason code:

Cause No. 47 - resource unavailable, unspecified.
This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies.

Since there is no codec or codec class or codec defined it will use G729 codec, can the other side accept g729? Can you added codec class listing both G711 and G729 to see if that works?

Alos, I see the dialed number as 178XXXXXXXX, did you mask it? What is the translation profile on the dial peers suppose to do?

[Pablo Escribano Aparicio]

Thanks for the reply.

I have the codec list already configured. Inside the dial peer it is configured the line "voice-class codec 1". And the voice class is configured as follows with G711 and G729.

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729br8
 codec preference 4 g729r8

Do you want me to change the preference order of de codecs?

In regarding to the dialed number, yes, it was masked by me. 

Attached is the translation-rule configuration:

translation-rule 210
 Rule 1 ^1787 1787

Regards¡¡¡

[Nadeem Ahmed]

Can you please post the complete show run from GW. and also collect below debug while making a test call.

Debug h225 asn1

debug h245 asn1

debug ip tcp transcation

debug voice ccapi inout

Br,

Nadeem