> > About once a day I have noticed a phantom incoming call with a caller ID of > > [EMAIL PROTECTED]<cut off>. When I answer the call there is a dial tone > > and the call is disconnected. Any clues? > > > > David Koski > David and List, > I am having the same problem. > I have an * box at my house with 1 zap (pstn on a X100p clone from digit > networks) channel and one sip(linksys ATA). I am getting ring on the ATA but > there is no call comming in from the pstn. The following is the CLI output > when this happens. I know that there is no call on the pstn because i have > an "emergency phone"(frequent power outages) still connected to the PSTN > parallel to the * box and it never rings. All the SIP stuff is on an internal > lan only. I only call out on PSTN since all I have available here in > nowheare land is dial up :-( All work flawlessly except for this one > problem. > > - Starting simple switch on 'Zap/1-1' > Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 > (Ring/Answered)... > -- Executing Dial("Zap/1-1", "sip/677|35") in new stack > -- Called 677 > -- SIP/677-55a8 is ringing > == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' > -- Hungup 'Zap/1-1' > Is there anything for Zap like sip debug? My first guess is that I am getting > some sort of blip in ring voltage on the PSTN but have no way to prove this. > As a posible logic check I unplugged from PSTN, which put zap into Red alarm > of course, and then i get no phantom calls. Is there something in the zap > driver that shuts down when in red alarm? > Any Ideas?
Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users