Bind port

Name/username Host Dyn Forcerport ACL Port Status Description GUS_iphone/GUS_iphone D N 32980 OK (130 ms)

I have bindport=5060 in sip.conf, but still my client does not use that port. Any help?

Beyond help. The client is the only thing that chooses the port that it uses at its end.

What problem are you trying to solve?

the question is: “what does Asterisk do, when I specify bindport=5060”

bindport = Number : UDP Port to bind to (listen on). Used to be port in Asterisk v1.0.x. Default 5060.

Asterisk 11

(You can choose independently for UDP, TCP, and TLS, by specifying different values for
"udpbindaddr", “tcpbindaddr”, and “tlsbindaddr”.)
(Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)

You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
for TLS).
IPv4 example: bindaddr=
IPv6 example: bindaddr=[::]:5062

udpbindaddr= ; IP address to bind UDP listen socket to ( binds to all)
; Optionally add a port number, (default is port 5060)

Thank you! However, I am still not able to solve my problem. So, i try again:

How can I configure Asterisk to make sure, that only sip port 5060 is allowed. So, no client can connect to any other port, that 5060.

That is the default behaviour. The port being reported in your original post is the port used on the peer. SIP has no control over that.

Again, what is your real problem. The diagnostic output indicates that SIP is working.

One thing to note is that you will need to open RTP ports, as well as SIP ones. Maybe your problem is that you have only opened 5060 and are wondering why you are getting no audio.