1 Nov
2008
1 Nov
'08
9:36 p.m.
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(a)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