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

[Larry Moore]

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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