Re: [Freeswitch-users] Buffer size sanity check failed!

[Rodrigo Andrés Zumaeta]

Ken,
thanks for your reply, let's see:
1. there is a group of users using analog phones.
2. there is another group using digital phones.
3. there is a third group that is not relevant for this specific problems.

All of these groups are connected to a Nortel, and we get H.323 traffic from them. The first group of users, the ones using analog phones, is the one reporting this problem. It happens with 100% of their calls. I am attaching a trace and console log... The only noticeable difference in the flow of the call (comparing to successful calls from digital phones) is the presence of CN RTP packets.




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On Feb 10, 2012, at 6:17 PM, Ken Rice wrote:

> Can you provide more information on that group of users? Can you duplicate
> this at will? ie: is it a specific hard phone or softphone using a specific
> codec?
> 
> Also console logs and network traces would be helpful...
> 
> 
> On 2/10/12 1:24 PM, "Rodrigo Andrés Zumaeta" <rzumaeta at yahoo.com> wrote:
> 
>> Hello, this is my first message to the list!
>> when connecting one specific group of users to Freeswitch, I get this message:
>> 
>> [CRIT] switch_core_codec.c:760 Buffer size sanity check failed!
>> 
>> The call is then dropped. That is, on 1.0.6. When I upgrade to the latest git
>> version, I still get the same message, but the call is not dropped anymore.
>> However, I get only one way audio.
>> 
>> As aforementioned, I just get this odd behavior with one group of users.
>> 
>> I will be happy to provide any relevant information.
>> 
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
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>> 
>> 
>> 
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> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
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