[OpenSIPS-Users] trouble with mediaproxy and asterisk on callforwarding (not 302)

[Uwe Kastens]

Hi,

I am using opensips 1.5.3 and the latest mediaproxy. I have a strange
issue with asterisk servers. Maybe somebody might have a hint where to
look.

The problem is reproducable with any asterisk server behind nat
(nat=yes). Calls in and out are working without any problem, RTP is
present, ringback is correct.

If I start a call to the asterisk server with an extension which makes a
dialout (call forwarding), which is simply a call in and a call out, the
calling side will hear no ringback and if the call is established, rtp
is send and received but could not be heared on both sides.



BR

Uwe


-- 

kiste lat: 54.322684, lon: 10.13586
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