[asterisk-users] call from h323 to SIP

[nik600]

Hi

i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.

Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.

h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x       ; this SHALL contain a single, valid IP
address for this machine
allow=all

extension.conf:
exten = 3298,1,Answer
exten = 3298,2,Dial(SIP/user at 193.y.y.y)

If a make a call to callamanager CISCO that forward to 3298 i read in
asterisk console:

Log:

Verbosity is at least 20
    -- Executing Answer("H323/ip$172.z.z.z:4836/14", "") in new stack
    -- Executing Dial("H323/ip$172.z.z.z:4836/14",
"SIP/user at 193.y.y.y") in new stack
    -- Called user at 193.y.y.y
    -- SIP/user at 193.y.y.y-40455d68 is ringing
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to ulaw
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec ........
.......
translation path from g729 to slin
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to ulaw
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to slin
Dec 15 14:45:13 WARNING[19794]: translate.c:116
ast_translator_build_path: No translator path from alaw to unknown
Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies:
Cannot build a path from g729 to slin
Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to
transmit frame type 64, while native formats is 256 (read/write =
4/64)
Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to
transmit frame type 256, while native formats is 4 (read/write = 4/4)
Dec 15 14:45:13 WARNING[19794]: translate.c:116
ast_translator_build_path: No translator path from alaw to unknown
Dec 15 14:45:13 WARNING[19794]: channel.c:2752
ast_channel_make_compatible: No path to translate from
H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8)
Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to
drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible
with SIP/193.x.x.x-40455d68
  == Spawn extension (default, 3298, 2) exited non-zero on
'H323/ip$172.z.z.z:4836/14'

Why? where am i wrong?