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

[Matteo Campana]

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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