Turn off callprogres=yes or have it configured properly. It should fix your problem.
regards Martin On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas <[email protected]> wrote: > Greetings listers. > > I’m running asterisk 1.4.21.2 on SUSE 11.0 using > Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. > For the most part I can make and receive calls fine except for these 3 > issues: > > 1. When I call an external conference, the call never bridges and > hangs up after 60-90 seconds. > > 2. When I call another number there is a 2-4 second delay before the > callee can hear me. > > 3. When I call an external conference and connect, the others cannot > hear me. > > > > Zapata.conf > > [trunkgroups] > > > > [channels] > > ;context=from-zaptel > > ;context=line1 > > busydetect=yes > > callprogress=yes > > busycount=4 > > hanguponpolarityswitch=yes > > answeronpolarityswitch=yes > > usecallingpres=yes > > priindication=outofband > > pritimer=t305,50000 > > signalling=fxs_ks > > wink=50 > > useincomingcalleridonzaptransfer=yes > > echocancel=yes > > echocancelwhenbridged=yes > > faxdetect=yes > > rxgain=1.0 > > txgain=21.0 > > callgroup=1 > > group=1 > > usecallerid=yes > > callerid=asreceived > > cidstart=ring > > hidecallerid=no > > immediate=no > > pickupgroup=1 > > ;context=incoming > > channel => 1-4 > > > > Sip.conf > > [general] > > srvlookup=yes ;allows DNS lookups of server names > > naxexpirey=180 > > defaultexpirey=160 > > context=default ; Default context for incoming calls > > allowoverlap=no ; Disable overlap dialing support. (Default is yes) > > bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) > > tos_sip=cs3 > > tos_audio=ef > > > > ; bindport is the local UDP port that Asterisk will > > ; listen on > > bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls > > limitonpeers=yes > > notifyringing=yes > > rtupdate=yes[authentication] > > > > [104] > > type=peer > > context=phones > > host=dynamic > > fromuser=104 > > secret=xxxxxx > > canreinvite=update > > directrtpsetup=no > > call-limit=3 > > nat=yes > > qualify=yes > > register=no > > session-timers=accept > > session-expires=90 > > session-minse=120 > > session-refresher=uac > > register => 104:[email protected]/104 > > defaultip=192.168.xx.xxx > > mailbox=104 > > disallow=all > > allow=ulaw,alaw > > artcachefriends=yes > > notifyhold=yes > > incominglimit=1 > > call-limit=3 > > > > Other information will be provided as asked for. > > > > Thanks in advance for any help you can provide. > > > > Danny Nicholas > > _______________________________________________ > -- 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 > _______________________________________________ -- 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
