I believe you're correct. And that should be the correct setting. However, You may want to do a packet sniff and confirm you're seeing the actual traffic as expected. Being that you see timeouts on the asterisk side. My bet is the rtp/sip traffic is going toward the device on a port it's not expecting. Or, The NAT device doesn't have a mapping for and being dropped at one of your routing devices.
Nick Olsen Network Operations (855) FLSPEED x106 ---------------------------------------- From: "John Millican" <[email protected]> Sent: Thursday, January 02, 2014 11:07 AM To: [email protected] Subject: Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: > Make sure you have nat=yes in your sip.conf either under globals or > individual sip peer settings. > > Nick Olsen > Network Operations > (855) FLSPEED x106 > > > > ------------------------------------------------------------------------ > *From*: "John Millican" <[email protected]> > *Sent*: Thursday, January 02, 2014 10:50 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > *Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> > NAT/Firewall-> Asterisk > > Hello, > CentOS 6.x and Asterisk 11.x > I have an interesting, to me at least, situation. Using a Polycom > 501(also tried with X-Lite). I have set up Asterisk to accept > registration from the Polycom and it registers successfully but then > withing 30 seconds on the CLI I get the message that the Polycom is > unreachable. The phone still shows that it is registered and if I try > to place a call from the phone to my Cell, my cell rings once and then > stops. I get a packet retransmission error: > WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout > reached on transmission [email protected] for seqno 2 (Critical > Response) > Followed by: > n_sip.c:4203 retrans_pkt: Hanging up call [email protected] - no > reply to our critical packet > I am "assuming" that there is a problem with NAT. I have externip set > in sip.conf. > Any pointers to what I am missing? > Thanks, > JohnM > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
