[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

[Ricky Gutierrez]

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 at 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 at 192.168.10.1:5070>;tag=as42edbc9b 
To: <sip:113 at 192.168.10.1>;tag=6d45d2188218c8ef 
Call-ID: 5a1c60382f18bd832fa3bdc54dc6ab13 at 192.168.10.1 
CSeq: 102 INVITE 
User-Agent: Grandstream GXP2020 1.1.6.16 
Contact: <sip:113 at 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


      
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