28 Sep
2005
28 Sep
'05
9:39 a.m.
Hi again, I made the same test with mediaproxy instead of rtpproxy, the
problem is exactly the same.
In my config there is no NAT at all, private IP addresses are routed via
an IP tunnel to my SER proxy, and I'd like to use the media proxy to
communicate with an external gateway. The SDP within the 200OK is left
intact, so a private IP address is given and the call fails...
Any ideas?
Thanks!
Le mardi 27 septembre 2005 à 18:30 +0200, Alexandre
Aractingi a écrit :
--
Alexandre Aractingi <aaractingi(a)libertysurf.fr>
Hi all, I'm trying to use RTPproxy (CVS) and
SER to enable
locally-routed endpoints to talk to an external public gateway. When a
call comes from the external gateway, the SDP in the INVITE gets
rewritten properly with SER's IP address, but then when my phone hangs
off, the SDP in the 200 OK doesn't get rewritten, so an RFC1918 IP
address is passed to the external gateway (so I get no audio).
I'm using nathelper with:
modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
and for my tests I put "force_rtp_proxy()" at the very beginning of the
routing logic in order to make sure it gets invoked.
But the SDP in the 200 OK is not rewritten no matter what. Is this a
know issue? Is there a known workaround?
Thanks in advance for any help,
Alexandre Aractingi
Net-tone / Active Telecom
Direct IP : +33 1 72 74 70 02
Standard : +33 1 49 23 76 59 
28 Sep
28 Sep
9:56 a.m.
Alexandre Aractingi wrote:
Hi again, I made the same test with mediaproxy instead
of rtpproxy, the
problem is exactly the same.
In my config there is no NAT at all, private IP addresses are routed via
an IP tunnel to my SER proxy, and I'd like to use the media proxy to
communicate with an external gateway. The SDP within the 200OK is left
intact, so a private IP address is given and the call fails...
You have to set a reply-route and use the mediaproxy/rtpproxy there too
for 180/183/200 having an SDP body.
Check the GettingStarted document on http://onsip.org for example
configurations using mediaproxy or rtpproxy.
Andy
10:38 a.m.
New subject: [Serusers] Sems IVR Module
Hi all,
I am testing the sems version from the Head and when i uses a SIP Phone to
call into
the IVR module for recording when I am in the middle of recording, the
program detected DTMF
signal and jump out of the script. Why is it so? as I never pressed on any
key? How can I resolve this
problem? Why is the voice signal become the DTMF signal? Is there any
adjustment that
i can do to solve this problem , as my voice goes louder the more DTMF
signal is being detected.
Please help.
regards,
nicky
11:07 a.m.
Le mercredi 28 septembre 2005 à 15:56 +0200, Andreas Granig a écrit :
You have to set a reply-route and use the
mediaproxy/rtpproxy there
too
for 180/183/200 having an SDP body.
Check the GettingStarted document on http://onsip.org for example
configurations using mediaproxy or rtpproxy.
Thanks for your answer. Actually my ser.cfg is derived from a simple
onsip.org example. The weird thing is that I have an on-reply_route and
a t_on_reply("1") statement, and still I don't see the call coming in
the onreply-route logs (even when the 200 OK is sent). Maybe that
explains the problem.
I attached my config, is there any obvious problem there?
Thanks a lot for your kind help,
--
Alexandre Aractingi <aaractingi(a)libertysurf.fr>
29 Sep
29 Sep
7:34 a.m.
I had a similar problem. When I took a look at SDP part of the
message, the IP of the proxy was always 127.0.0.1.
In mediaproxy.ini, I put:
[MediaProxy]
proxyIP = my_IP
and it worked. Now mediaproxy was putting a valid IP on the SDP body.
Felipe
--
Master Student - Electrical Engineering Department
Computer Engineering and Telecommunications Research Group
Universidade Federal de Minas Gerais - Brazil
"For God so loved the world that he gave his one and only Son, that
whoever believes in him shall not perish but have eternal life."
John 3:16
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4 comments
4 participants
participants (4)
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Alexandre Aractingi -
Andreas Granig -
Felipe Louback -
Nicky