We had that problem but changing busydetect from on to off fixed it. It appears that you already have that covered.
-- -- Steven http://www.glimasoutheast.org "Doug Lytle" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > Hey everybody, > > I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've > received several complaints about dropped calls. > Reviewing the archives on PRI and dropped calls shows that I should set the > resetinterval=never in the zapata.conf and restart. > This hasn't helped. > The dropped calls have to date only been on outbound calls. Usually within 2 > to 3 minutes of a call. The full log shows > something about not getting a frame and stopping the bridge. > > On Saturday I put into place 1.2 Branch and have pri debug setup to log to a > file. Is there anything else that I can do to get an > idea as to what is going on here? > > My zapata and zaptel below: > > [zaptel] > > # Zaptel Configuration File > > span=1,1,0,esf,b8zs > defaultzone=us > loadzone=us > bchan=1-23 > dchan=24 > > span=2,0,0,esf,b8zs > fxsks=25-32 > fxoks=33-48 > defaultzone=us > loadzone=us > > [zapata] > > [channels] > ; > context=default > resetinterval = never > musiconhold=tape > > switchtype=national > context=pri > signalling=pri_cpe > group=1 > echocancel=yes > echotraining=yes > echocancelwhenbridged=yes > rxgain=-1.0 > txgain=-2.0 > busydetect=no > pridialplan=unknown > usercallerid=yes > callerid=asreceived > channel => 1-23 > > I see the following the full log: > > Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Executing > Dial("SIP/4228-082131e8", "ZAP/G1/1xxxxxx5800") in new stack > Oct 4 09:09:30 DEBUG[29894] dsp.c: dsp busy pattern set to 0,0 > Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Requested transfer > capability: 0x00 - SPEECH > Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Called G1/xxxxxx5800 > Oct 4 09:09:30 VERBOSE[29894] logger.c: -- Zap/23-1 is proceeding > passing it to SIP/4228-082131e8 > Oct 4 09:09:32 VERBOSE[29894] logger.c: -- Zap/23-1 is ringing > Oct 4 09:09:37 VERBOSE[29894] logger.c: -- Zap/23-1 answered > SIP/4228-082131e8 > Oct 4 09:11:26 DEBUG[29894] channel.c: Didn't get a frame from channel: > SIP/4228-082131e8 > Oct 4 09:11:26 DEBUG[29894] channel.c: Bridge stops bridging channels > SIP/4228-082131e8 and Zap/23-1 > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO MODE, value: ON(1) > on Zap/23-1 > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Hangup: channel: 23 index = 0, > normal = 40, callwait = -1, thirdcall = -1 > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Not yet hungup... Calling hangup > once with icause, and clearing call > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo cancellation on > channel 23 > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Set option TDD MODE, value: OFF(0) > on Zap/23-1 > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Updated conferencing on 23, with 0 > conference users > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: Set option AUDIO MODE, value: OFF(0) > on Zap/23-1 > Oct 4 09:11:26 DEBUG[29894] chan_zap.c: disabled echo cancellation on > channel 23 > Oct 4 09:11:26 VERBOSE[29894] logger.c: -- Hungup 'Zap/23-1' > Oct 4 09:11:26 DEBUG[29894] app_dial.c: Exiting with DIALSTATUS=ANSWER. > Oct 4 09:11:26 VERBOSE[29894] logger.c: == Spawn extension (sip, > xxxxxxxxx5800, 5) exited non-zero on 'SIP/4228-082131e8' > Oct 4 09:11:26 VERBOSE[29894] logger.c: -- Executing > NoOp("SIP/4228-082131e8", "Hungup") in new stack > Oct 4 09:11:26 VERBOSE[29894] logger.c: -- Executing > Hangup("SIP/4228-082131e8", "") in new stack > > > -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase > a little Temporary Safety, deserve neither Liberty > nor Safety." > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
