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 :
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
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
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
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,
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