Yes ... as a matter of fact here is the sip.conf ... obviously private info removed .... [general] register => 1747xxxxxxx:[email protected]<1747xxxxxxx%[email protected]> port = 5060 bindaddr = 192.168.22.5 context = incoming svrlookup=yes ;dtmfmode=inband allow=all externip=76.98.xxx.xxx localnet=192.168.22.0/255.255.255.0
[proxy01.sipphone.com] nat=yes ;type=peer type=friend context=incoming disallow=all allow=ulaw allow=alaw allow=ilbc dtmfmode=rfc2833 host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com ;insecure=very deprecated; use insecure=port,invite instead insecure=port,invite qualify=yes secret=XXXXXXXXXXXX authuser=1747XXXXXXX fromuser=1747XXXXXXX username=1747XXXXXXX canreinvite=no On Wed, Aug 5, 2009 at 6:07 AM, Administrator TOOTAI <[email protected]>wrote: > Rob a écrit : > > Hi all, > > > Hi > > I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a > > while and it works fine .... I just added CALL OUT ... I have no problem > > with call setup ... the called party hears me ... but I can't hear them > .... > > again if the call comes INTO the server both sides work fine. > > > > > Looks like a nat issue: do you have nat=yes and canreinvite=no in your > sip.conf for Gizmo5? > > -- > Daniel > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
