[asterisk-users] The call is established but without exchanged voice packets

[Matthew J. Roth]

Asmaa Ahmed wrote:
> 
> 
> I am trying to make my first call on Asterisk to succeed. I have
> Asterisk 1.8.10.1 running on Ubuntu machine. 
> 
> The configuration is quite simple just for my first test, Trying to
> have a call between two X-lite sipphone. The subscribers succeeded
> to register and the call is established, but still no voice can be
> heard, a nd lead the call to be disconnected after! By checking the
> logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission
> timeout reached on transmission
> Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2
> (Critical Response) 

The SIP trace you provided breaks down as follows:

  X-Lite               Asterisk
  ---------------      -------------------------------
  INVITE(No Auth) ---> 
                  <--- 401 Unauthorized
  ACK             --->
  INVITE(Auth)    --->
                  <--- 100 Trying
                  <--- 200 OK
                  <--- 200 OK (Retransmitted 10 Times)
                  <--- BYE
  OK              --->

This shows that the three-way handshake (INVITE/200 OK/ACK) used to
establish SIP sessions is not completed because Asterisk never
receives an ACK from X-Lite.  After retransmitting the 200 OK 10 times
Asterisk gives up and disconnects the call.

> Here's my simple sip configuration 
> [general] 
> context=internal 
> allowguest=no 
> allowoverlap=no 
> bindport=5060 
> bindaddr=0.0.0.0 
> srvlookup=no 
> disallow=all 
> allow=ulaw 
> alwaysauthreject=yes 
> canreinvite=no 
> nat=yes 
> session-timers=refuse 
> externip=<IP>