[Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

Michael Jerris mike at jerris.com
Fri Dec 11 09:51:13 PST 2009


As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on.

Mike

On Dec 11, 2009, at 11:44 AM, Chris Chen wrote:

> Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly
> I noticed that the acl automatically having 192.168.0.0 set as "deny", that's why I tried to changed the settings regarding nat acl and localnet acl.
> 
> Chris
> 
> 
> 
> On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle <frank at carmickle.com> wrote:
> On Fri, Dec 11, Chris Chen wrote:
> > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN
> > trunk 15905.
> 
> Is this a change in behavior or is this the first time you've run freeswitch?  If this is your first time welcome aboard!  Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled.  This was the behavior I saw the first time I ran it on a box with aliases on an interface.  The stun server tells freeswitch after some time that the IP is different then the one you've assigned.  This is just one possibility.  If this isn't the case then we will need to see sip traces on all of your profiles.
> 
> --FC
> 
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