I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk:
allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is behind a Linksys router (full cone NAT). About 25% of the time, when I call that number (from another sipphone account), asterisk answers the line, but about 75% of the time, asterisk fails to answer, and doesn't even indicate that any incoming call was attempted, and sipphone times out after 15-20 seconds and dumps the unanswered call to its voicemail system. I don't see any pattern to the intermittent answering, and sometimes I can try numerous times and get no answer, and sometimes I can try several times in a row and get an answer each time. It seems random. Outgoing calls work 100%; only incoming are having problems. How can I diagnose whether the problem is with Asterisk or with Sipphone, or whether one or both are having problems because of NAT? Bypassing the NAT router is not an option, even for testing. Is this a known problem with Sipphone? How do the various voip providers (Sipphone, FWD, Broadvoice, etc) compare with regards to incoming call completion reliability when the receiving device (Asterisk in this case) is behind NAT? I'll eventually need to accept incoming PSTN calls via voip and I'm willing to pay for reliable service from any provider, but I do need Asterisk to actually receive and answer all attempted incoming calls. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
