On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote: > We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10, > acting as gateways from SIP to ISDN PRI interfaces. Each has one > Digium TE420 (with hardware echo cancellation) and one TC400B for > transcoding, in order to handle 60/90 contemporary calls in peak > hours. > > In my logs there are hundreds of thousand warnigs per day like these: > > transcode.c: no samples for lintoulaw > transcode.c: zapg729toalaw did not update samples ### > > Could you suggest me what are the possible causes for that? Are they > signs of bad audio quality? Any ideas for resolving these issues? > > In addition I can say that we are using a quite large jitter buffer in > zapata.conf: > > jitterbuffers=16 (=> 0.32s) > > Moreover, it uses the fixed implementation, because when I tried the > adaptive one I experienced one-way audio. > Finally I have to note that, using a Siemens IP phone (G.729 no > AnnexB) in conditions of no load on servers, I could replicate > non-deterministically (sigh!) each of these problems, with a very > noisy audio, and a annoying period of silence during the first seconds > of call. > > Regards, > Francesco > > PS. Previous versions of asterisk and zaptel presented an identical situation. > Have you tried additional types of phones and if so can you produce the same non-deterministic problems?
Dave _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
