[SR-Users] Call from SipML5 -> PSTN

[Jeremy Jongepier]

On 11/28/2012 08:58 PM, Konstantin M. wrote:
> Jeremy, it is doesn't work at all. I've made a lot of changes to that
> patched asterisk to make it working and no luck.
> However, ast11 has fully supported webrtc, but I heard no voice during a
> call.
> Another issue is - sipml5 is sending a malformed Contact field, and
> asterisk is trying to contact to invalid destination and finally closing a
> call.

Hello Konstantin,

Thanks for the heads up. Those sound like issues that could be resolved. 
No audio or one way audio is almost always either a codec or a NAT issue 
and the malformed Contact field is something I think could be worked 
around too.

Regards,

Jeremy