sip set debug peer 90102 and check in log why call drop or upload log somewhere. configuration seems ok.
On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk <[email protected]>wrote: > Current configuration follows: > > [general] > context=default > allowguest=no > alwaysauthreject=yes > allowoverlap=yes > allowtransfer=yes > tcpenable=no > tlsenable=no > srvlookup=yes > vmexten=vm > rtcachefriends=yes > nat=no > directmedia=nonat > directrtpsetup=no > videosupport=yes > maxcallbitrate=384 > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > allow=ilbc > allow=speex > allow=g726 > allow=g723 > mohinterpret=default > mohsuggest=default > dtmfmode=rfc2833 > timer1b=60000 > transport=udp > > [carrier-1] > host=a.b.c.d > type=friend > context=from-pstn > disallow=all > allow=ulaw,alaw > qualify=yes > trunk=yes > > [90102] > secret=xxxxxx > mailbox=90102@default > cid_number=NXXXXXXXXX > accountcode=401 > type=friend > host=dynamic > port=5060 > qualify=yes > nat=yes > transport=udp > context=users > disallow=all > allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263 > directmedia=no > canreinvite=no > videosupport=no > >
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