Config H323 between CUCM and GW 2900

Pablo Escribano Aparicio · ‎08-26-2015

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 · ‎08-26-2015

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 · ‎08-26-2015

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 · ‎08-27-2015

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

Br, Nadeem Please rate all useful post.