Hi, I’m having trouble configuring Asterisk with my trunk provider and it’s driving me nuts.
my local network is located behing the firewall. Asterisk is part of the local network.
The phone I created and the trunk are registered but for some reason outbound calls do not work.
The necessary ports are opened on the firewall and natted to the asterisk server.
but every time I try to make a call i have this message :
<— SIP read from
SIP/2.0 401 Unauthorized
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5,
xx.xx.xx.xx being my WAN address
yy.yy.yy.yy being the SIP Provider address
The SIP Provider is using port 5026 (declared using port=5026 in the sip.conf file)
everything seems to be ok.
Dumping the packets with wireshark confirms port 5026 is used.
My provider told me this was a nat problem but i’ve tried any configuration possible for the nat value nothing seems to work. He also told me rport=1024 was the problem.
Has anyone got an idea about what is causing rport to be 1024 no matter what I try (nat=force_rport,nat=comedia or both, nat=no, nat=yes I’ve tried them all)