Hello there,
I have set a simple environment to test some functionalities of
asterisk's new jitterbuffer.
The environment is composed of a sip softphone registering in asterisk
1.4 and calling a pstn phone connected to asterisk through a fxs
board.
Using the fixed buffer implementation the call quality is improved
when injecting a artificial jitter in my local network. However, when
changing to the adaptive buffer and making the same call in the same
environment, no audio is received in my phone and I get a warning in
asterisk console: "abstract_jb.c: Failed to put first frame in the
jitterbuffer on channel ZAP".
This is my zapata.conf:
[trunkgroups]
[channels]
context=from-pbx
signalling=fxo_ks
usecallerid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
busydetect=yes
busycount=6
channel=>1
jbenable=yes
jbimpl=adaptive
Has anyone experienced such things? Any tips?
Thanks in advance, folks.
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