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