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

2009-12-11 Thread Brian West
Please test www.bkw.org/sofia_autonat_static_ip.diff

/b

On Dec 11, 2009, at 1:34 PM, Chris Chen wrote:

> Thanks Mathieu, but I am on SVN r15912 now.
>
> Chris


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

2009-12-11 Thread Chris Chen
Thanks Brian for your explanation, could we still keep the option to set the
extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe
many other routers have similar issue.
Chris


On Fri, Dec 11, 2009 at 2:45 PM, Brian West  wrote:

> You set the extrtp ip to an IP exactly.. this is the issue we are
> fixing soon.. if you have natpmp or upnp set it to auto-nat and let it
> figure it out.  The issue is we have restored the behavior in 1.0.4
> that lies about the IP all the time...
>
> I'm going to commit a patch shortly that'll fix this.
>
> /b
>
> On Dec 11, 2009, at 1:34 PM, Chris Chen wrote:
>
> > Thanks Mathieu, but I am on SVN r15912 now.
> >
> > Chris
>
>
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Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Brian West
You set the extrtp ip to an IP exactly.. this is the issue we are  
fixing soon.. if you have natpmp or upnp set it to auto-nat and let it  
figure it out.  The issue is we have restored the behavior in 1.0.4  
that lies about the IP all the time...

I'm going to commit a patch shortly that'll fix this.

/b

On Dec 11, 2009, at 1:34 PM, Chris Chen wrote:

> Thanks Mathieu, but I am on SVN r15912 now.
>
> Chris


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

2009-12-11 Thread Chris Chen
Thanks Mathieu, but I am on SVN r15912 now.

Chris

On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene  wrote:

> Its not sending to the right Contact:  header in the 200 OK packet. This
> was fixed in r15870, you have to update.
>
>  Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mr...@avgs.ca
>
>
>
>
> On 11-Dec-09, at 2:03 PM, Chris Chen wrote:
>
> 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  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 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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Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Mathieu Rene
Its not sending to the right Contact:  header in the 200 OK packet.  
This was fixed in r15870, you have to update.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca




On 11-Dec-09, at 2:03 PM, Chris Chen wrote:

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

2009-12-11 Thread 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  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 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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Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Michael Jerris
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  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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Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Chris Chen
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 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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Re: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks

2009-12-11 Thread Frank Carmickle
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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