Hi list, always looking for as solving my audio problem with mediaproxy asterisk and openser, there will be some form of telling to the openser that when he comes from the from sip:asterisk@192.168.10.1:5070 that doesn't use the mediaproxy or the onreply_route[1] ,

SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK36b7f619;rport=5070 
Record-Route: <sip:192.168.10.1;lr=on;ftag=as42edbc9b;nat=yes> 
From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as42edbc9b 
To: <sip:113@192.168.10.1>;tag=6d45d2188218c8ef 
Call-ID:
 5a1c60382f18bd832fa3bdc54dc6ab13@192.168.10.1 
CSeq: 102 INVITE 
User-Agent: Grandstream GXP2020 1.1.6.16 
Contact: <sip:113@192.168.10.30:5062;transport=udp> 
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE 
Content-Type: application/sdp 
Supported: replaces, timer 
Content-Length: 212 
P-hint: Onreply-route - fixcontact  
P-hint: onreply_route|usemediaproxy  
 
v=0 
o=113 8000 8000 IN IP4 192.168.10.30 
s=SIP Call 
c=IN IP4 192.168.1.64 
t=0 0 
m=audio 35064 RTP/AVP 0 101 
a=sendrecv 
a=rtpmap:0 PCMU/8000 
a=ptime:20 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 

 making other tests if I change this form the onreply_route[1], I have audio in the openser extension, but the one that this behind the pstn doesn't have audio or he doesn't listen to me

onreply_route[1] {
        #
        #-- On-replay block routing --
        #
        if (client_nat_test("1")) {
            append_hf("P-hint: Onreply-route - fixcontact \r\n");
            fix_nated_contact();
        };

        if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) {
            if (search("^Content-Type:[ ]*application/sdp")) {
             append_hf("P-hint: onreply_route|usemediaproxy \r\n");
               use_media_proxy();
           };
       };
       exit;
}

my best regardss
rickygm