I am using Asterisk 13.6.0, and I have a SIP TRUNK with an ISP.
The calls are hangup in the middle of the call, the ISP told me that I have to active on the INVITE support to the 100rel, PRACK and UPDATE.
I dont know how to do that, at moment my sip.conf is like that:
[general] context=interno allowguest=no bindport=5060 binaddr=0.0.0.0 language=pt_BR disallow=all allow=alaw dtmfmode=rfc2833 nat=force_rport,comedia limitonpeer=yes srvlookup=no rtptimeout=60 rtpholdtimeout=300 tos_sip=cs3 tos_audio=ef tos_video=af41 t38pt_udptl=yes ignoresdpversion=yes [voxip] type=peer host=10.150.66.16 outboundproxy=10.150.66.16 fromuser=5430468800 insecure=invite,port qualify=yes port=5060 nat=no disallow=all allow=g729,alaw prack=yes reinvite=yes canreinvite=no username=5430468800 secret=5430468800 context=entrada-voxip rtptimeout=10
I didn’t see the option prack on the
sip.conf, but the ISP told me that I have to include this on my
The ISP sent me an image showing that my Asterisk isn’t sent the UPDATE, what is needed to keep the call going on otherwise the ISP drop the call.