I have no problem doing that by adding the information you have under [general] to /etc/asterisk/sip_nat.conf
On Thu, Jan 29, 2009 at 7:30 AM, Holger Latz <t...@globalview.de> wrote: > > Hi all, > > I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone > in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the > phone is ringing, but when I pickup the call, there's no audio on both > sides. > > I debugged the rtp-traffic at home. As long as the phone is ringing, > everything is fine. But after the pickup, asterisk sends a SIP/SDP > package with its private address (192.168.100.10). After the softphone > received this package, it tries to send RTP data to this address! > Obviously those packages never reach asterisk... > > Does 'externip' just works for SIP and not for RTP? > Where does the the internal IP-address come from and how can I set the > right one? > > > My configuration: > > [general] > externip = 85.XXX.XXX.XXX > nat = yes > localnet = 192.168.100.0/24 > > [42] > deny=0.0.0.0/0.0.0.0 > disallow=all > type=friend > secret=XXX > qualify=yes > port=5060<http://0.0.0.0/0.0.0.0disallow=alltype=friendsecret=XXXqualify=yesport=5060> > pickupgroup= > permit=0.0.0.0/0.0.0.0 > nat=yes > mailbox...@device > host=dynamic > dtmfmode=rfc2833 > dial=SIP/42 > context=from-internal > canreinvite=no<http://0.0.0.0/0.0.0.0nat=yesmailbox...@devicehost=dynamicdtmfmode=rfc2833dial=sip/42context=from-internalcanreinvite=no> > callgroup= > callerid=device <42> > allow=alaw > accountcode= > call-limit=50 > > > Regards > Holger > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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