[Asterisk-Users] incoming calls dropout on PRI over TE110p

[Paul C]

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ).   

I have upgraded libpri and zaptel to trunk, but I don't want to upgrade Asterisk to 1.2 until I've got this all sorted,  one problem at a time!

Here are my configs :

/etc/zaptel.conf

# Global data

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17
dchan=16
unused=18-31
loadzone = au
defaultzone = au

/etc/asterisk/zapata.conf

[channels]
context=from-onramp

overlapdial=yes
priindication = outofband
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe

echocancelwhenbridged=yes
echocancel=128
echotraining=800

rxgain=5 ; 0
txgain=-4.5 ; 0
busydetect=no
pridialplan=local
internationalprefix=0011
nationalprefix=0
usecallerid=yes
hidecallerid=no
callprogress=no

group=0
channel => 1-15,17


/etc/asterisk/extensions:

[from-onramp]
;exten => s,1,Playback(custom/aa_1)
exten => s,1,Dial(SIP/116)
exten => h,1,Hangup

and here's some log info:

asterisk*CLI> pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&    -- Going to extension s|1 because of Complete received
    -- Executing Dial("Zap/1-1", "SIP/116") in new stack
    -- Called 116
    -- Accepting call from '' to 's' on channel 0/1, span 1
    -- SIP/116-5a95 is ringing
&&&&&&&&&&&&&&&&&    -- SIP/116-5a95 answered Zap/1-1
  == Spawn extension (from-onramp, s, 1) exited non-zero on 'Zap/1-1'
    -- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (from-onramp, h, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'

and going straight to a Playback command rather than SIP extension:

asterisk*CLI> pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%    -- Going to extension s|1 because of Complete received
    -- Executing Answer("Zap/2-1", "") in new stack
    -- Accepting call from '' to 's' on channel 0/2, span 1
  == Spawn extension (from-onramp, s, 1) exited non-zero on 'Zap/2-1'
    -- Executing Hangup("Zap/2-1", "") in new stack
  == Spawn extension (from-onramp, h, 1) exited non-zero on 'Zap/2-1'
    -- Hungup 'Zap/2-1'




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