<div>maybe you can try to issue a "sip show registry" on the console on a regular basis and watch if your * loose registration.</div>
<div>You can also turn on sip debug on the console, to see if the "unanswered calls" effectively reach asterisk or not. In the latter, is sipphone that loose your registration, so you maybe can lower the time before registration renewals. And turn on "qualify=yes" for your peer to keep fresh nat mappings on the router. Search
<a href="http://voip-info.org">voip-info.org</a> for more infos....</div>
<div> </div>
<div>Hope this helps....<br> </div>
<div><span class="gmail_quote">2006/4/27, jnuoiqweahf kajhdsff <<a href="mailto:jnuoiqweahf@yahoo.com">jnuoiqweahf@yahoo.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I have a <a href="http://sipphone.com">sipphone.com</a> account, with asterisk set to<br>answer incoming calls, using the following settings
<br>(phone number and password omitted) in the Peer<br>Details for the SIP Trunk:<br><br>allow=ulaw<br>context=from-pstn<br>dtmfmode=rfc2833<br>fromdomain=<a href="http://proxy01.sipphone.com">proxy01.sipphone.com</a><br>
fromuser=1747xxxxxxx<br>host=<a href="http://proxy01.sipphone.com">proxy01.sipphone.com</a><br>insecure=very<br>secret=xxxxx<br>type=peer<br>username=1747xxxxxxx<br><br>The Asterisk machine is behind a Linksys router (full
<br>cone NAT).<br><br>About 25% of the time, when I call that number (from<br>another sipphone account), asterisk answers the line,<br>but about 75% of the time, asterisk fails to answer,<br>and doesn't even indicate that any incoming call was
<br>attempted, and sipphone times out after 15-20 seconds<br>and dumps the unanswered call to its voicemail system.<br>I don't see any pattern to the intermittent answering,<br>and sometimes I can try numerous times and get no
<br>answer, and sometimes I can try several times in a row<br>and get an answer each time. It seems random. Outgoing<br>calls work 100%; only incoming are having problems.<br>How can I diagnose whether the problem is with
<br>Asterisk or with Sipphone, or whether one or both are<br>having problems because of NAT? Bypassing the NAT<br>router is not an option, even for testing. Is this a<br>known problem with Sipphone? How do the various voip
<br>providers (Sipphone, FWD, Broadvoice, etc) compare<br>with regards to incoming call completion reliability<br>when the receiving device (Asterisk in this case) is<br>behind NAT?<br><br>I'll eventually need to accept incoming PSTN calls via
<br>voip and I'm willing to pay for reliable service from<br>any provider, but I do need Asterisk to actually<br>receive and answer all attempted incoming calls.<br><br>__________________________________________________<br>
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