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>
>From the SIP trace, I believe 'externip=41.46.164.96' is set. If that
is the case, try changing it to 'externip=54.241.129.14'. You should
also set localnet as follows:
; RFC 1918 addresses
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
If that doesn't work you can also try setting 'nat=force_rport'
instead of 'nat=yes'.
> [7001]
> type=friend
> host=dynamic
> secret=123
> context=internal
>
> [7002]
> type=friend
> host=dynamic
> secret=456
> context=internal
>
> A snoop capture for my call is uploaded in the following link. I
> wonder if there is any missing configuration or plugin need to be
> set here!
> http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992
>
At this point, you should be able to establish a call between the two
X-Lite phones that won't get disconnected due to failing to complete
the three-way handshake. There may still not be voice because the
firewall(s) between Asterisk and the X-Lite phones may block the RTP
traffic. The phones appear to be on the same network, so you can try
setting 'canreinvite=yes' to workaround this problem until the
firewall(s) are configured to allow RTP traffic on the UDP port range
specified in 'rtp.conf' (the default range is 10000-20000).
Good luck and please report your progress back to the list.
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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