Hi,I am experiencing a strange issue. I am forwarding
calls to Asterisk voicemail upon no-response. So, in
failure route, I simply check the status 408 or 486
and if true, I jump to a routing-block. In there, I
have the following:
-----
revert_uri();
rewritehost(Asterisk-IP);
append_branch();
if (isflagset(6) || isflagset(7)) use_media_proxy();
if (!t_relay()...)
------
Now, Asterisk does receive the invite, except that the
SDP in the SIP invite looks like (PAY ATTENTION TO c=
field).
-------------------------------
v=0
o=208500512 8000 8001 IN IP4 192.168.1.104
s=SIP Call
c=IN IP4 213.189.X.Y213.189.X.Y
t=0 0
m=audio 3532235322 RTP/AVP 0 8 4 18 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20
------------------------------------------
The c= field has the same IP twice and thus the INVITE
is rejected by Asterisk.
Is this a bug and did someone have such an issue?
Thanks
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