update again and reproduce the sanity check err and send the new value now with more info. On Fri, Feb 10, 2012 at 3:57 PM, Rodrigo Andrés Zumaeta <rzumaeta@yahoo.com> wrote:
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.
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@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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