[asterisk-users] PRI issues

[Doug Lytle]

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


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