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@voipstunt
-- Got SIP response 302 "Moved temporarely" back from
194.120.0.203
-- Now forwarding mISDN/1-1 to
'SIP/+497121479628@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