Re: [Freeswitch-users] Buffer size sanity check failed!
Rodrigo Andrés Zumaeta
rzumaeta at yahoo.com
Sat Feb 11 00:57:48 MSK 2012
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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> http://www.freeswitchsolutions.com
>
>
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