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The problem is only with PSTN call.<BR>
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I tried to send call to the three gateway Teles, SIP-HIT and Asterisk but all disconnect calls in that priticular seconds.<BR>
The thinng is i cannot understand if i am using STUN in Linksyspap2 the call goes normal and without STUN it disconnect. So the problem is gateway side or Openser?<BR>
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our router is not implimented with SIP, and there is one more strange thing, In some callshop the same rtptproxy working well and going cal for long duration but i have 3 callshop which facing this problem. the configuration and others are same as other working devices.<BR>
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<BR><BR>Regards, <BR>www.Go4Calls.Com <BR>VoIP Forums <BR><BR>> From: ibc@aliax.net<BR>> To: users@lists.openser.org<BR>> Date: Mon, 21 Jan 2008 00:19:12 +0100<BR>> Subject: Re: [OpenSER-Users] Calls disconnect automatically<BR>> <BR>> El Lunes, 21 de Enero de 2008, VoIP Forums www.Go4Calls.com escribió:<BR>> > i tired with the following configuration but still result is same. calls<BR>> > disconnect in 30 - 32 sec<BR>> ><BR>> > modparam("nathelper", "natping_interval", 20)<BR>> > modparam("nathelper", "ping_nated_only", 1)<BR>> > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")<BR>> > modparam("nathelper", "rtpproxy_disable", 0)<BR>> > modparam("nathelper", "rtpproxy_disable_tout", 60)<BR>> > modparam("nathelper", "rtpproxy_tout", 1)<BR>> > modparam("nathelper", "rtpproxy_retr", 5)<BR>> > modparam("nathelper", "sipping_method", "OPTIONS")<BR>> > modparam("nathelper", "received_avp", "$avp(i:801)")<BR>> ><BR>> > Please advise me if i need more modification?<BR>> <BR>> <BR>> Which kind of calls are disconnected after 30 seconds? PSTN calls or user to <BR>> user call?<BR>> <BR>> In any case, you could do a "tcpdump -n port UAC_RTP_PORT" in a PC using a <BR>> softphone that uses UAC_RTP_PORT for audio. Call to PSTN (or other user) from <BR>> this softphone and monitorize with tcpdump when the audio is disconnected.<BR>> <BR>> Some gateways (as Asterisk) disconnect a call by default if they don't receive <BR>> RTP during 30 seconds.<BR>> <BR>> Since I don't know which kind of gateway you use I don't know if it uses <BR>> Session Timers as call monitorization way, so if your router blocks the port <BR>> after 30 seconds, then the periodic ire-INVITE or UPDATE from gateway to UAC <BR>> will not arrive so they won't be replied with "200 OK", and gateway will <BR>> discconect the call.<BR>> To test this, do a "ngrep" in a computer using a softphone registered behind <BR>> NAT (no STUN). After REGISTER you should receive a OPTIONS from proxy as keep <BR>> alive.<BR>> <BR>> Another possible problem is the existence of painful ALG routers, have you <BR>> tested if your router implements SIP ALG?<BR>> <BR>> <BR>> <BR>> <BR>> -- <BR>> Iñaki Baz Castillo<BR>> <BR>> _______________________________________________<BR>> Users mailing list<BR>> Users@lists.openser.org<BR>> http://lists.openser.org/cgi-bin/mailman/listinfo/users<BR><BR><br /><hr />Express yourself instantly with MSN Messenger! <a href='http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/' target='_new'>MSN Messenger</a></body>
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