[asterisk-users] No Audio

[Wasif]

Hello,

My DIDs are hitting Directly to Asterisk Machine via SIP G729. From there I
am forwarding call to Cisco 3845 via SIP G729. And from Cisco calls are
terminating to my carriers via H323 G729.

DIDs------------> Asterisk ----------> Cisco 3845 --------> Carrier
	Sip G729	 	     Sip G729		   H323 G729

Cisco is capable to convert calls from SIP to H323 and H323 to SIP.
My Problem is when a call hits to my Carrier I get no audio at all. Other
side gets ring but upon answering there is no audio.

Below is the output from CLI. I have installed G729 codec in system and its
working fine; there is no Firewall and NAT implementation in my scenario. 

Called Cisco3800/99999999999999
Destroying call '0541efff076643ca2cab9c800fe8b2e2 at Y.Y.Y.Y'
asterisk1*CLI>
<-- SIP read from X.X.X.X:50974:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK47392650;rport
From: "4169076956" <sip:4169076956 at Y.Y.Y.Y>;tag=as2db17260
To: <sip:99999999999999 at X.X.X.X>;tag=4B93E20-172
Date: Fri, 28 Jul 2006 20:13:13 GMT
Call-ID: 47c75ec474fe9df50eff04c50a7d34bf at Y.Y.Y.Y
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

<-- SIP read from X.X.X.X:50974:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK47392650;rport
From: "4169076956" <sip:4169076956 at Y.Y.Y.Y>;tag=as2db17260
To: <sip:99999999999999 at X.X.X.X>;tag=4B93E20-172
Date: Fri, 28 Jul 2006 20:13:13 GMT
Call-ID: 47c75ec474fe9df50eff04c50a7d34bf at Y.Y.Y.Y
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:99999999999999 at X.X.X.X:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 261

v=0
o=CiscoSystemsSIP-GW-UserAgent 5381 5741 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 18068 RTP/AVP 18 101
c=IN IP4 X.X.X.X
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port X.X.X.X:18068
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263
|h263p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
    -- SIP/Cisco3800-056c is making progress passing it to SIP/5060-08e53008


After few seconds call gets disconnect. 


x.x.x.x  Asterisk Box 
y.y.y.y Cisco 3845 12.3T

[Cisco3845]
;disallow=all
allow=g729
dtmfmode=auto
host=y.y.y.y
insecure=very
sendrpid=yes
type=friend


Thanks