On 01/28/2012 10:22 AM, Din Assegaf wrote:
> Hi All,
> I’m trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
> But when making A Call from SIP Client, I got cli Warning … and no
> call has been made.
> My Sip Client is using lib java peers client http://peers.sourceforge.net/
> with standard codec PCMU/PCMA
> [Jan 28 23:03:32] WARNING: chan_sip.c:8942 process_sdp:
> Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101
> [Jan 28 23:03:32] WARNING: chan_sip.c:9029 process_sdp: Failing
> due to no acceptable offer found
> the strange thing is when using asterisk 1.6, is normal,
> when using asterisk 1.8.x and using another client like Ekiga is normal too,
The error message is misleading; you are having this problem because the
‘m’ line in the SDP with the ‘audio’ offer has a port number of 0
(zero)., which means it is not an active media stream offer. It does not
make any sense for the SDP in an INVITE for a new call to have an m-line
with a port number of zero.
I’ll improve the error message so that this sort of situation won’t be
as confusing in the future.