I am attempting to setup a SIP phone that is behind NAT router, to hook up to my Asterisk server the phone is a Grandstream BudgetTone100 has anyone had any luck doing this. Thanks Doug D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041108/2c45ddde/attachment.htm
Doug L. Dawson wrote:> I am attempting to setup a SIP phone that is behind NAT router, to hook > up to my Asterisk server the phone is a Grandstream BudgetTone100 has > anyone had any luck doing this. >add the following to sip.conf for the grandstream's context host=dynamic nat=yes also look at http://www.voip-info.org/wiki-Asterisk+config+sip.conf for more information. it should be clearly explained there in the "SIP configurations - peers and clients" section. cheers flynn
Hi all, I am setting up a a proof on concept where a SIP phone sits on the internet and connects to a * behing a NAT. Right now the SIP phone connects to the * box just fine, I can dial and I see the commands being executed on the * box, but I don't have any audio on the SIP phone. Any idas/pointers? Thanks, Andre Courchesne
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