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

Reply via email to