[Asterisk-Users] Cut off after 8 secs?? Help

[Rich Adamson]

> Call comes in from remote SIP, authorised, does the following and dies
> 
> Any idea why..
> 
> I have ports 5060 and 16384 to 16482 open
> 
> Do I need any others?
> 
> What am I missing
> 
> Remote user is using X-lite for windows..
>  
>     -- Executing Dial("SIP/2004-944c", "SIP/2001|20") in new stack
>     -- Called 2001
>     -- SIP/2001-4f3b is ringing
>     -- SIP/2001-4f3b answered SIP/2004-944c
>     -- Attempting native bridge of SIP/2004-944c and SIP/2001-4f3b
> Jul  5 19:04:17 WARNING[-224801872]: chan_sip.c:497 retrans_pkt: Maximum
> retries exceeded on call
> FC7F4A8C-3CC8-42C6-BF13-892EA5CC6503 at 192.168.1.100 for seqno 24922
> (Response)

It is dying because the audio stream (rtp packets) aren't getting
through. Not sure why you picked rtp ports 16384-16482; each sip phone
vendor picks there own set of port ranges, and the Xlite product use
to use ports in the 8000 range (haven't checked lately).

Read the stuff on the wiki relative to NAT parameters (for *) and
you should be able to get it to work.