lots of warnings from translate.c

1 msg	Cisco 79X1 speaker issue
4 msg	Asterisk sends 486 Busy Here instead of 600 Bus...
1 msg	Dear asterisk-users April 84% 0FF
4 msg	WARNING: Remote host can't match request NOTIFY...
1 msg	Check the answered channel in simultaneous sip ...
40 msg	Quality problems with ISDN PRI

lots of warnings from translate.c
\ Francesco Castellano (22 Apr 2008)
. \ David Boyd (22 Apr 2008)
. . \ Francesco Castellano (22 Apr 2008)

3 msg	features.conf Problem with DTMF_sequence
6 msg	Parsing incoming extension till first @
3 msg	OT: Linksys devices send incorrect REGISTER
9 msg	Conditional include=> ?
1 msg	Strange SIP packet
1 msg	caller groups
13 msg	Can I roll my own E911?
9 msg	Disable transfer on all calls
9 msg	Switch recommendation?
2 msg	Monitor v/s MixMonitor
3 msg	Phone notification?
3 msg	Dual Interface config
6 msg	Monitor not merging calls

Subject:	lots of warnings from translate.c
Group:	Asterisk-users
From:	Francesco Castellano
Date:	22 Apr 2008

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.

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