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

[Chris Chen]

Hi Mike, the fs console log with sip trace on the internal profile is
attached in the pastebin below,
http://pastebin.freeswitch.org/11483

could you please take a look at it?
Thanks,
Chris

On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris <mike at jerris.com> wrote:

> 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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