Can you show us how the previous INVITE Looked like vs the current one? *José Pablo Méndez *********
On Sun, Jan 1, 2012 at 4:17 PM, <cov...@ccs.covici.com> wrote: > 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. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > cov...@ccs.covici.com > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
