RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

[Manny A. Wise]

Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT…. if you make it work just by opening ports, let me know..I have never been able to get it to work, that’s why I don’t use sip, just plain iax2 for everything… J   Manny   -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath Sent: Wednesday, November 23, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Thanks Michael, I think thats is the problem, I have opened only ports 5060-5082, I need to open 10000-20000 as well. I will try that and post the result when i get back home. Thanks On 11/23/05, Michael West <[EMAIL PROTECTED]> wrote: I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 10000-20000 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 10000-20000) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom ---------------------------------------------------------- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bharath Khambadkone Sent: Wednesday, November 23, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make & recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? Thanks On 11/22/05, C F <[EMAIL PROTECTED]> wrote: On 11/22/05, Bharath Khambadkone <[EMAIL PROTECTED]> wrote: > Hello All, >  I'm fairly new to asterisk. I have read about the problems about NAT, But > can't seem to find a solution. >  My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? > the asteris box. I have sucessfully connected iax2 softphones & was able to > recieve & make calls. In the same locations where I have the iax2 extensions > working I have set up a a SIP softphone & a SIP ATA (Sipura2002). Both teh > sip phones are able to register. I can also make & recieve calls but cannot > hear anything after the call is answered at both ends. I'm not sure what is > causing this problem. By the way I'm using SME server 7(centos 4.2)  with > [EMAIL PROTECTED] installed. > >  my Sip.conf : >  [2008] ;(Sipura2002) >  username=2008 >  type=friend >  secret=2008 >  record_out=Adhoc >  record_in=Adhoc >  qualify=no >  port=5060 >  nat=1 >  [EMAIL PROTECTED] >  host=dynamic >  dtmfmode=rfc2833 >  context=from-internal >  canreinvite=no >  callerid=device <2008> > > >  [2009] ;X-Lite Soft Phone >  username=2009 >  type=friend >  secret=2009 >  record_out=Adhoc >  record_in=Adhoc >  qualify=no >  port=5060 >  nat=1 >  [EMAIL PROTECTED] >  host=dynamic >  dtmfmode=rfc2833 >  context=from-internal >  canreinvite=no >  callerid=device <2009> > >  Thanks in advance..

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