Hi,
Maybe you can check the contact of INVITE from sip proxy to callee.
Try fix_contact () for INVITE.
Regards,
Lesley
2007/6/4, Andreas Granig <agranig@sipwise.com>:
Hi,
I'm going crazy with this problem, so maybe anyone here can see any
error or can give a hint how to debug this.
I've two clients behind (the same) NAT. INVITE goes thru, so does 100
and 180. But the 200 isn't relayed because of the following syslog error:
./sbin/openser[23933]: ERROR: udp_send:
sendto(sock,0x81816a8,814,0,0xb5b7f168,16): Operation not permitted(1)
./sbin/openser[23933]: msg_send: ERROR: udp_send failed
I can't find any error (the two Via headers are exactly the same), so
here is the 180 which is relayed without problems, and below the 200,
which causes the error:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 84.19.176.11;branch=z9hG4bK6351.1b30ad43.0
Via: SIP/2.0/UDP
192.168.123.100:2083;received=
83.65.24.161;branch=z9hG4bK-kq6qyyeowiei;rport=12499
From: "Foo Bar" <sip:foo1000@linguin.org>;tag=xuu6a2tp6b
To: <
sip:2000@linguin.org;user=phone>;tag=2091138726
Call-ID: 3c27e5729c40-csdi3p954zyn@snom360-0004132306CE
CSeq: 2 INVITE
Server: Cisco ATA 186 v3.1.1 atasip (040629A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.19.176.11;branch=z9hG4bK6351.1b30ad43.0
Via: SIP/2.0/UDP
192.168.123.100:2083
;received=83.65.24.161;branch=z9hG4bK-kq6qyyeowiei;rport=12499
Record-Route: <sip:84.19.176.11;lr=on;ftag=xuu6a2tp6b;nat=yes>
From: "Foo Bar" <
sip:foo1000@linguin.org>;tag=xuu6a2tp6b
To: <sip:2000@linguin.org;user=phone>;tag=2091138726
Call-ID: 3c27e5729c40-csdi3p954zyn@snom360-0004132306CE
CSeq: 2 INVITE
Contact: <sip:foo2000@192.168.123.103:5060;transport=udp>
Server: Cisco ATA 186 v3.1.1 atasip (040629A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Supported: replaces
Content-Length: 204
Content-Type: application/sdp
v=0
o=foo2000 11828 11828 IN IP4 192.168.123.103
s=ATA186 Call
c=IN IP4 192.168.123.103
t=0 0
m=audio 16384 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Any ideas?
Regards,
Andreas
PS: this happens with both OpenSER 1.1 and
1.2
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