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:
[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()
[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 (template) username=9250 secret=password mailbox=9250 (template) username=9251 secret=password mailbox=9251