Hi. I am using asterisk 1.8 and everything was working fine when I was
at svn 342661. I then upgraded to vrsion 349339 and discovered the
following problem — one of the end points is a freeswitch box which
offers a number of codecs, including PCMU. However, when I tried to
make a call I got a 488 response and a message “multiple audio streams
not supported” in the log.
Is this by design? I found an issue 18859, but that referenced where
the end point offered both regular rtp and srtp. But it seems to me if
an endpoint offers various codecs, that asterisk could only complain if
none of them match one that asterisk likes.
If I only offer one codec, it works, but that seems an unnecessary
restriction to me.
Any assistance on this would be appreciated.