[Asterisk-Users] Problem: have no RTP streams from Asterisk

[Leutin Alexandr]

Good day. 
I'm trying to configure termination with The Asterisk thru Cisco
AS5300 Gateway from the SIP softphone (X-Ten X-Lite) to POTS network.
I think, I had recognise kind of problem: call is ringing in the 
POTS phone (so I guess SIP signalling is working ok?), but there is
no voice in either sides.
On the Asterisk PC I can see incoming RTP streams with tcpdump and 
tethereal, but I can't to see any RTP outgoing streams.
There is G.711 a-law codec selected on Asterisk and Cisco, and I
see it in sniffed RTP's.

How can I fix this trouble? 

I have attached configuration files's pieces.

Best regards, Alexandr.
-------------- next part --------------
[203]
type=friend
host=dynamic
username=203
secret=nastya
nat=no
canreinvite=no
context=office
callerid="Nastya"  <203>
mailbox=203 at office
qualify=1000
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[ciscoout]
type=peer
host=176.16.1.16
canreinvite=no
qualify=1000
disallow=all
allow=alaw

-------------- next part --------------
[office]
exten => 8,1,SetCallerID(2393030)
exten => 8,2,Dial(SIP/73512606002 at ciscoout,30,rtTwW)