asterisk users - Jun 2011 - No audio after a reinvite changing codec ----> with SIP DEBUG!!

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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On 28/06/2011 6:59 PM, Matteo Campana wrote:
>
>
> 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
>
>
You mention you have an SPA962, I expect the configuration will be the 
same if not similar to an SPA942. It would be worth checking what your 
"Symmetric RTP" setting is, you can find it listed in the RTP
Parameters
section under the SIP section of your phone 
http://<ip_address_of_spa962>/admin/advanced.

If it is set to "no" set it to "yes".

Larry.
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On Fri, Jul 1, 2011 at 12:05 PM, Larry Moore <lmoore at starwon.com.au>
wrote:
> **
> On 28/06/2011 6:59 PM, Matteo Campana wrote:
>
>
>
>  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
>
>
>
> You mention you have an SPA962, I expect the configuration will be the same
> if not similar to an SPA942. It would be worth checking what your
"Symmetric
> RTP" setting is, you can find it listed in the RTP Parameters section
under
> the SIP section of your phone http://
> <ip_address_of_spa962>/admin/advanced.
>
> If it is set to "no" set it to "yes".
>
> Larry.
>
>
Hy Larry,
I have tested with "Symmetric RTP = yes" in SPA962, but with same
results.

Matteo
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