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hi:<br>how about the codecs? <br><br><br>Best wishes!<br>Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support.<br>website:www.cnasterisk.com, www.voip88.com<br><br><br><br><br>> Date: Wed, 31 Mar 2010 17:20:30 -0500<br>> From: brent@texascountrytitle.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] Dropped Calls<br>> <br>> On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:<br>> ><br>> > Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?<br>> > <br>> I was suspecting something with either rtptimeout or sip registration <br>> timeout, but I'm not sure what.<br>> <br>> -- <br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> <br /><hr />Hotmail: Trusted email with powerful SPAM protection. <a href='https://signup.live.com/signup.aspx?id=60969' target='_new'>Sign up now.</a></body>
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