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. -- []'s André de Abrantes _______________________________________________ -- 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
