<div id="RTEContent">please help!!!<br>  <br>  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?<br>  <br>  thanks<br>  <br>  sip.conf:<br>  [general]<br>  context=local-access&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  ; Default context for incoming calls<br>  bindport=5060&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  ; UDP Port to bind to (SIP standard port is 5060)<br>  bindaddr=0.0.0.0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  ; IP address to bind to (0.0.0.0 binds to all)<br>  srvlookup=yes&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;  ; Enable DNS SRV lookups on
 outbound calls<br>  musicclass=default<br>  <br>  [authentication]<br>  <br>  [1000]<br>  username=1000<br>  regexten=1000<br>  mailbox=1000@voicemail<br>  callerid="jon Smith" &lt;1000&gt;<br>  context=local-access<br>  nat=yes<br>  secret=password<br>  type=friend<br>  host=dynamic<br>  canreinvite=yes<br>  disallow=all<br>  allow=all<br>  <br>  [1001]<br>  username=1001<br>  regexten=1001<br>  mailbox=1001@voicemail<br>  callerid="jane doe" &lt;1001&gt;<br>  context=local-access<br>  nat=yes<br>  secret=password<br>  type=friend<br>  host=dynamic<br>  canreinvite=yes<br>  disallow=all<br>  allow=all<br>  <br>  extensions.conf:<br>  [general]<br>  static=yes<br>  writeprotect=no<br>  autofallthrough=yes<br>  clearglobalvars=no<br>  priorityjumping=no<br>  <br>  [globals]<br>  ATTENDANT=1001<br>  OUTBOUNDTRUNK=ZAP/g1<br>  <br>  [extentions]<br>  exten =&gt; _10XX,1,Ringing<br>  exten =&gt; _10XX,2,Dial(SIP/${EXTEN},20)<br>  exten =&gt; _10XX,3,Answer<br>  exten =&gt;
 _10XX,4,VoiceMail(u${EXTEN}@voicemail)<br>  exten =&gt; _10XX,5,Hangup<br>  <br>  [voicemail]<br>  exten =&gt; _910XX,1,Wait(1)<br>  exten =&gt; _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)<br>  <br>  [local]<br>  include =&gt; extentions<br>  include =&gt; voicemail<br>  <br>  [incoming]<br>  exten =&gt; s,1,Answer<br>  exten =&gt; s,2,Background(our-voicemail-sound)<br>  exten =&gt; t,1,Playback(vm-goodbye)<br>  exten =&gt; t,2,Hangup( )<br>  exten =&gt; 0,1,Dial(SIP/${ATTENDANT},20)<br>  exten =&gt; 1,1,Directory(voicemail,internal,f)<br>  exten =&gt; 2,1,Directory(voicemail,internal)<br>  include =&gt; extentions<br>  <br>  [local-trunks]<br>  exten =&gt; _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})<br>  exten =&gt; _9XXXXXXXXXX,2,Congestion( )<br>  exten =&gt; _9XXXXXXXXXX,102,Congestion( )<br>  exten =&gt; 911,1,Dial(${OUTBOUNDTRUNK}/911)<br>  exten =&gt; 9911,1,Dial(${OUTBOUNDTRUNK}/911)<br>  <br>  [local-access]<br>  ignorepat =&gt; 9<br>  include =&gt; local<br> 
 include =&gt; local-trunks<br>  <br>  <br>  zapata.conf:<br>  <br>  [trunkgroups]<br>  [channels]<br>  context=default<br>  switchtype=national<br>  signalling=fxo_ls<br>  rxwink=300&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ; Atlas seems to use long (250ms) winks<br>  usecallerid=yes<br>  hidecallerid=no<br>  callwaiting=yes<br>  usecallingpres=yes<br>  callwaitingcallerid=yes<br>  threewaycalling=yes<br>  transfer=yes<br>  canpark=yes<br>  cancallforward=yes<br>  callreturn=yes<br>  echocancel=yes<br>  echocancelwhenbridged=yes<br>  rxgain=0.0<br>  txgain=0.0<br>  group=1<br>  callgroup=1<br>  pickupgroup=1<br>  immediate=no<br>  group=1<br>  echocancel=yes<br>  switchtype=national<br>  signalling=fxs_ks<br>  context=incoming<br>  echocancelwhenbridged=yes<br>  channel =&gt; 1-4<br>  <br>  <br>  /etc/zaptel.conf:<br>  fxsks=1,2,3,4<br>  loadzone = us<br>  defaultzone=us<br>  <br>  log:<br>  Asterisk Ready.<br>  &nbsp;&nbsp;&nbsp; -- Starting
 simple switch on 'Zap/1-1'<br>  Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18 (Ring Begin)...<br>  Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie made mylen &lt; 0 (-155)<br>  Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID feed failed: Success<br>  Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID returned with error on channel 'Zap/1-1'<br>  &nbsp;&nbsp;&nbsp; -- Executing Answer("Zap/1-1", "") in new stack<br>  &nbsp;&nbsp;&nbsp; -- Executing BackGround("Zap/1-1", "our-voicemail-sound") in new stack<br>  &nbsp;&nbsp;&nbsp; -- Playing 'our-voicemail-sound' (language 'en')<br>  &nbsp; == CDR updated on Zap/1-1<br>  &nbsp;&nbsp;&nbsp; -- Executing Ringing("Zap/1-1", "") in new stack<br>  &nbsp;&nbsp;&nbsp; -- Executing Dial("Zap/1-1", "SIP/1000|20") in new stack<br>  &nbsp;&nbsp;&nbsp; -- Called 1000<br>  &nbsp;&nbsp;&nbsp; -- SIP/1000-54e4 is ringing<br>  &nbsp;&nbsp;&nbsp; -- SIP/1000-54e4 answered
 Zap/1-1<br>  &nbsp; == Spawn extension (incoming, 1000, 2) exited non-zero on 'Zap/1-1'<br>  &nbsp;&nbsp;&nbsp; -- Hungup 'Zap/1-1'<br>  <br>  </div><p>
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