[asterisk-users] Dealing with FINAREA redirects

[Arik Raffael Funke]

Hi,

does anybody currently use voipstunt from finarea? I place a call to 
sip.voipstunt.com I get a 302 redirection. Unfortunately the second 
server seems to support only a different set of codecs than the first:

     -- Called +497121479628 at voipstunt
     -- Got SIP response 302 "Moved temporarely" back from 194.120.0.203
     -- Now forwarding mISDN/1-1 to 
'SIP/+497121479628 at 80.239.235.201:5060' (thanks to SIP/voipstunt-081c1ba0)
Sep 14 15:36:56 WARNING[12025]: chan_sip.c:2561 sip_write: Asked to 
transmit frame type 8, while native formats is 4 (read/write = 4/4)

My question: How to I get asterisk to re-negotiate the codecs with the 
new handler? - Or am I interpreting something wrong here?

Regards,
Arik