[asterisk-users] No audio after a reinvite changing codec ----> with SIP DEBUG!!

[Matteo Campana]

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <lmoore at starwon.com.au> wrote:

> On 18/06/2011 5:36 AM, Matteo Campana wrote:
>
>>
>> Inviato da iPhone
>>
>> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com>
>>  ha scritto:
>>
>>  We experience the same thing.  The solution we use is to not change
>>> codecs in the middle of a call.   I assumed it was an issue with our
>>> upstream.
>>>
>>
>> Hi Eric,
>> this behavior  is an asterisk bug or asterisk can never change the codec
>> "on the fly"?
>>
>>
>> Thanks,
>> Matteo
>>
>>
> The problem reported occurs after a fax tone is detected, one might expect
> T.38 or G711 to be used to handle the fax, not G729!
>
> There is no configuration file information for each of the nodes/peers, no
> debug of each peer involved  nor a trace of the packets hence no one will
> have ideas!
>
> Larry.



Hi Larry,
I have the SIP debug taken from asterisk.
In this debug:     1.2.3.4 ---> IP SIP PROXY
                         5.6.7.8 ---> IP UAC (Linksys SPA 962)
                         9.10.11.12 ---> IP ASTERISK to connect to the
provider
                         13.14.15.16 --> IP PROVIDER
                         17.18.19.20 --> IP ASTERISK


The SIP debug is available at this link: http://pastebin.com/9DrFDmeC


Thanks in advance,
Matteo










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