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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5070>;tag=as42edbc9b
To: <sip:[EMAIL PROTECTED]>;tag=6d45d2188218c8ef
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:[EMAIL PROTECTED]: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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