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...00001 DCall: 00001 [192.168.0.25:4569] -- Hungup 'IAX2/iax1Centos-1'CentOS*CLI> *****************************************************************************************;CentOS server iax.conf[general]bindport=4569bindaddr=192.168.0.22 ; this is Centos server address [iax1Centos]type=friendsecret=volgausername=iax1Centoscontext=centos-contexthost=dynamictrunk=yes--------------------;Centos server extension.conf[centos-context]exten =>3000,1,Set(TIMEOUT(absolute)=10000) exten =>3000,2,dial(sip/3000/20)exten =>3000,3,voicemail(u3000 at default)exten=> _2XXX,1,Answer()exten=> _2...

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