I am getting Error 483, too many Hops, There is no other error messages i am getting. Please some one help me out in this
From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Rajesh Babu Sent: Friday, 27 September, 2013 6:08 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork HI Mike, Now the RTP is up and i am getting this message on my logs [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]: INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17815]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17808]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17811]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17814]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:10.10.10.123:7890> found, support for it enabled Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]: WARNING:drouting:dr_load_routing_info: table "dr_rules" is empty Sep 28 01:59:43 centos64 opensips: INFO:core:daemonize: pre-daemon process exiting with 0 But my test tool is not connecting back my server. Is there any mistake i am doing. Thanks Rajesh From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Rajesh Babu Sent: Friday, 27 September, 2013 2:34 PM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork Hi Mike, This is log i am geting wheni try to start the service [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]: WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has been disabled temporarily Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy Connection refused Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not respond, disable it Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has been disabled temporarily Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy Connection refused Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not respond, disable it Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has been disabled temporarily Sep 27 22:29:03 centos64 opensips: INFO:core:daemonize: pre-daemon process exiting with 0 From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Mike Tesliuk Sent: Thursday, 26 September, 2013 10:25 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork When you use the residential script almost all configuration come alredy working for this i have a tutorial (in portuguese ( i think that i should translate to english :) )) , where you can see a routing script working with nat http://opensips.com.br/wiki/index.php?title=Opensips_1.9 You can take a look at modules documentation too http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html There is on this maillist too a lot of discussions about this, below you can see one case http://opensips.org/pipermail/users/2011-January/016130.html If you get some information from an old version of opensips probably will be necessary to take a look on the module documentation to check about little diferences , but i think that this is the start point :) and if you is new to opensips i recommend to you the book about opensips ( http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book ) 2013/9/26 Rajesh Babu <rajesh.b...@goodcoresoft.com> Hi Mike, Thanks for the response, I am totally new to this world, can you please help me by directing to on how to configure links. It will be great. Thanks in advance Regards Rajesh From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Mike Tesliuk Sent: Thursday, 26 September, 2013 12:25 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork you should configure the nathelper and rtpproxy, this should help in you issue. 2013/9/26 Rajesh Babu <rajesh.b...@goodcoresoft.com> Hi, I am new to the OpenSIP world. I have installed a OpenSIP on my network. If i make a Call inside the network between two users i don't have any issue, where as from outside the network, even though i can see the user registered in my server i am not able to call registered user (I see the user in my UL show listing). The call is established but i am not able to talk (Mean the audio and video are not getting transffered). Where as messages are going fine without any issue. I guess it is because message transmit over XMPP where calls on SIP right. I am really struck and i don't know how to proceed, please help me out Thanks Rajesh _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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