Hello guys, I am having some problems with calls comming from the PSTN lines, when somebody calls people can't hear me, but I can hear them, every day I have to do a /etc/init.d/asterisk stop && /etc/init.d/dahdi restart to have calls with sound again, wich cli dubug commands can I use to see what is going on, here I have my chan_dahdi.conf and sip.conf, I am using 1.6 Thanks a lot! chan_dahdi.conf trunkgroups] [channels] Group=1 context=incoming signalling=fxs_ks rxwink=300 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes jbenable=no jbmaxsize=200 jbresyncthreshold=1000 useincomingcalleridondahditransfer=yes ;callerid=asrecived rxgain=0.0 txgain=0.0 immediate=no busydetect=yes busycount=5 hidecallerid=no callgroup=1 pickupgroup=1 channel => 1-24 sip.conf [general] disallow=all allow=gsm allow=ulaw language=es [sets](!) type=friend secret=1000 host=dynamic ;call-limit=1 [1109](sets) mailbox=1109 at default context=oficina [1110](sets) mailbox=1110 at default context=oficina [1111](sets) mailbox=1111 at default context=oficina-jefatura [1112](sets) mailbox=1112 at default context=oficina [1113](sets) mailbox=1113 at default context=oficina [1114](sets) mailbox=1114 at default context=oficinaVoIP [1115](sets) mailbox=1115 at default context=oficina [1116](sets) mailbox=1116 at default context=oficina [1117](sets) mailbox=1117 at default context=oficinaVoIP -- http://celord.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081111/a996be21/attachment.htm