Hello,
I’m trying to use the Playback function but my SIP Client (x-lite) do not receive any sound from asterisk server.
The RTP packets flow, coming from my SIP Client to Asterisk server, is ok but the RTP flow from Asterisk Server to SIP client is not received by the client.
There is no Nat, Firewall or any network problem (I think, I hope 
).
tcpdump capture on asterisk server:
[ul]Client SIP -> Asterisk_Server :SIP Request :Invite sip:1000@Asterisk_Server
Asterisk Server -> Client SIP :[color=red]SIP Status:401 Unauthorized[/color][/ul]
Can anybody tell me what is this response “401 Unauthorized” from the Asterisk server ? And why the sound played by asterisk server is not played by SIP client ?
Thanks by advance! 
Here my conf files:
extensions.conf:
[general]
static=yes
writeprotect=yes
[bogon-calls]
[from-sip]
exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Hangup
exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Hangup
exten => 1000,1,Answer()
exten => 1000,n,Playback(a)
exten => 1000,n,Hangup()sip.conf:
[general]
context=internal
allowoverlap=no
port=5060
bindaddr=x.x.x.x (asterisk server ip)
srvlookup=yes
tos_sip=cs3
tos_audio=0xb8
allow=ulaw
allow=alaw
context=bogon-calls
allowguest=yes
[template](!)
type=friend
context=from-sip
host=dynamic
nat=no
canreinvite=no
[9250](template)
username=9250
secret=password
mailbox=9250
[9251](template)
username=9251
secret=password
mailbox=9251