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[1654]: 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[1654]: 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.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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