[Asterisk-Users] ZAP <--> sip(polycom301) can not hear each other

[sdgesa gaeharth]

please help!!!
  
  I am dialing into our asterisk server(TDM400p) from the psnt. I hear  our voicemail message and I press the extention 1000. The Polycom ip  phone in the office rings. I pickup but neither side can hear one  another. What have I done wrong?
  
  thanks
  
  sip.conf:
  [general]
  context=local-access                  ; Default context for incoming calls
  bindport=5060                    ; UDP Port to bind to (SIP standard port is 5060)
  bindaddr=0.0.0.0                 ; IP address to bind to (0.0.0.0 binds to all)
  srvlookup=yes                    ; Enable DNS SRV lookups on outbound calls
  musicclass=default
  
  [authentication]
  
  [1000]
  username=1000
  regexten=1000
  mailbox=1000 at voicemail
  callerid="jon Smith" <1000>
  context=local-access
  nat=yes
  secret=password
  type=friend
  host=dynamic
  canreinvite=yes
  disallow=all
  allow=all
  
  [1001]
  username=1001
  regexten=1001
  mailbox=1001 at voicemail
  callerid="jane doe" <1001>
  context=local-access
  nat=yes
  secret=password
  type=friend
  host=dynamic
  canreinvite=yes
  disallow=all
  allow=all
  
  extensions.conf:
  [general]
  static=yes
  writeprotect=no
  autofallthrough=yes
  clearglobalvars=no
  priorityjumping=no
  
  [globals]
  ATTENDANT=1001
  OUTBOUNDTRUNK=ZAP/g1
  
  [extentions]
  exten => _10XX,1,Ringing
  exten => _10XX,2,Dial(SIP/${EXTEN},20)
  exten => _10XX,3,Answer
  exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail)
  exten => _10XX,5,Hangup
  
  [voicemail]
  exten => _910XX,1,Wait(1)
  exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)
  
  [local]
  include => extentions
  include => voicemail
  
  [incoming]
  exten => s,1,Answer
  exten => s,2,Background(our-voicemail-sound)
  exten => t,1,Playback(vm-goodbye)
  exten => t,2,Hangup( )
  exten => 0,1,Dial(SIP/${ATTENDANT},20)
  exten => 1,1,Directory(voicemail,internal,f)
  exten => 2,1,Directory(voicemail,internal)
  include => extentions
  
  [local-trunks]
  exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
  exten => _9XXXXXXXXXX,2,Congestion( )
  exten => _9XXXXXXXXXX,102,Congestion( )
  exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)
  exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
  
  [local-access]
  ignorepat => 9
  include => local
  include => local-trunks
  
  
  zapata.conf:
  
  [trunkgroups]
  [channels]
  context=default
  switchtype=national
  signalling=fxo_ls
  rxwink=300              ; Atlas seems to use long (250ms) winks
  usecallerid=yes
  hidecallerid=no
  callwaiting=yes
  usecallingpres=yes
  callwaitingcallerid=yes
  threewaycalling=yes
  transfer=yes
  canpark=yes
  cancallforward=yes
  callreturn=yes
  echocancel=yes
  echocancelwhenbridged=yes
  rxgain=0.0
  txgain=0.0
  group=1
  callgroup=1
  pickupgroup=1
  immediate=no
  group=1
  echocancel=yes
  switchtype=national
  signalling=fxs_ks
  context=incoming
  echocancelwhenbridged=yes
  channel => 1-4
  
  
  /etc/zaptel.conf:
  fxsks=1,2,3,4
  loadzone = us
  defaultzone=us
  
  log:
  Asterisk Ready.
      -- Starting simple switch on 'Zap/1-1'
  Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18 (Ring Begin)...
  Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie made mylen < 0 (-155)
  Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID feed failed: Success
  Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID returned with error on channel 'Zap/1-1'
      -- Executing Answer("Zap/1-1", "") in new stack
      -- Executing BackGround("Zap/1-1", "our-voicemail-sound") in new stack
      -- Playing 'our-voicemail-sound' (language 'en')
    == CDR updated on Zap/1-1
      -- Executing Ringing("Zap/1-1", "") in new stack
      -- Executing Dial("Zap/1-1", "SIP/1000|20") in new stack
      -- Called 1000
      -- SIP/1000-54e4 is ringing
      -- SIP/1000-54e4 answered Zap/1-1
    == Spawn extension (incoming, 1000, 2) exited non-zero on 'Zap/1-1'
      -- Hungup 'Zap/1-1'
  
  

		
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