H323 to cucm to sip CUBE call flow issue

[Mohammed Al-odhari]

Hi every one,

i have cucm version 8.5 with the following call flow:

Panaosoic PBX-->H323Trunk-->CUCM-->SipTrunk-->(ge0/0)cisco CUBE(ge0/1)-->sip-ITSP(Sip provider)

1- calls from panasonic PBX to ip phones on CUCM= success

2- calls from ip phones on CUCM to Cube & ITSP = success

3- calls from panasonic PBX to ITSP= fails, busy tone after dialing the last digit of the number

4- calls from ITSP to ip phones of the cucm=success

5- calls from ITSP to panasonic PBX= calls established but no 2 way audio.

Routing is ok, and cube configuration is attached.

In CUBE, the sip bind control and media are binded to Ge0/1 only.

in CUCM, sip trunk are pointed to the Ge0/1 of the cube (port connected to sip provider) with no preference dtmf signaling, also MTP required unchecked.

Is there any parameter to check like in h323 gateway (Panasonic) added in the cucm or any other place to sove the call flow 3 and 5.

thanks.

[Jinto Alakkal]

Hello,

Do you see the calls dialed from the Panasonic PBX landing in the CUBE, can you make a test call and collect "debug ccsip messages" debugs from the CUBE and post it here, make sure that you are applying a "u all" after collecting its output, also let me know the calling and called number so that it will be easy for me to identify the call.

Thanks!

Jinto.

[Mohammed Al-odhari]

Thanks Jinto for your replay,

please refer to attached for debug call from Panasonic to ITSP, debug collected from the CUBE in which calling party is 1101 (panasonic phone) and called called no is 0096655551040

regards,