[Asterisk-Users] G729 troubles

[Anton V Kirichenko]

Hello, 
I've successfully installed Asterisk from last CVS   and configured it
for using with DLINK-DG104S  as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
 All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip messages, but I see strange   string at asterisks log:

NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame

I find similary posts at Asteris-Users mailing list, but don't find how
to resolve this trouble.  Is this a bug or some misconfiguration at my
configs ?

sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes 
transfer=yes   
line => aaln/1
line => aaln/2
line => aaln/3
line => aaln/4
extension.conf
[local]
ignorepat => 9
exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP

Some logs from Asterisk:

First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/1 at DLINK MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000

After that I enter phone number and sent call to sip server:

    -- Executing Dial("MGCP/aaln/1 at DLINK-0", "SIP/3632034 at IP.IP.IP.IP")
in new stack

INVITE sip:3632034 at IP.IP.IP.IP SIP/2.0
<skip>
v=0
o=root 16078 16078 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 18480 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
<skip>

Sip read:
SIP/2.0 183 Session Progress
<skip>
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap:  1 image udptl t38
a=sqn: 0
a=cdsc:  1 image udptl t38

After this message sometimes Asterisk make error message at log and drop
call:

  -- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/1 at DLINK-1
srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write =
256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame

Reliably Transmitting:
CANCEL sip:3632034 at IP.IP.IP.IP:5060 SIP/2.0

Sip read: 
SIP/2.0 487 Request Cancelled
....

--
Antonio