<div id="RTEContent">That is correct, The SIP phones are all on our  LAN. I changed the nat's to say no, but I still get the same problem.  Another thing, when I call out to the pstn from our local sip phones.  The same problem happens.&nbsp; The outid line rings, the person picks  p but no sounds.<br>  <br>  &nbsp;Any suggestions????<br>  <br>  thanks<br><br><b><i>Ken D'Ambrosio &lt;ken@jots.org&gt;</i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">  &gt;From your description, it sounds as if the SIP phones are local to the<br>Asterisk box.  If this is so, having "nat=yes" might be a problem.<br><br>-Ken<br><br>sdgesa gaeharth wrote:<br><br>&gt; please help!!!<br>&gt;<br>&gt; I am dialing into our asterisk server(TDM400p) from the psnt. I hear<br>&gt; our voicemail message and I press the extention 1000. The Polycom ip<br>&gt; phone in the office rings. I pickup but neither side can hear one<br>&gt;
 another. What have I done wrong?<br>&gt;<br>&gt; thanks<br>&gt;<br>&gt; sip.conf:<br>&gt; [general]<br>&gt; context=local-access                 ; Default context for incoming calls<br>&gt; bindport=5060                   ; UDP Port to bind to (SIP standard<br>&gt; port is 5060)<br>&gt; bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds<br>&gt; to all)<br>&gt; srvlookup=yes                   ; Enable DNS SRV lookup s on outbound<br>&gt; calls<br>&gt; musicclass=default<br>&gt;<br>&gt; [authentication]<br>&gt;<br>&gt; [1000]<br>&gt; username=1000<br>&gt; regexten=1000<br>&gt; mailbox=1000@voicemail<br>&gt; callerid="jon Smith" &lt;1000&gt;<br>&gt; context=local-access<br>&gt; nat=yes<br>&gt; secret=password<br>&gt; type=friend<br>&gt; host=dynamic<br>&gt; canreinvite=yes<br>&gt; disallow=all<br>&gt; allow=all<br>&gt;<br>&gt; [1001]<br>&gt; username=1001<br>&gt; regexten=1001<br>&gt; mailbox=1001@voicemail<br>&gt; callerid="jane doe" &lt;1001&gt;<br>&gt;
 context=local-access<br>&gt; nat=yes<br>&gt; secret=password<br>&gt; type=friend<br>&gt; host=dynamic<br>&gt; canreinvite=yes<br>&gt; disallow=all<br>&gt; allow=all<br>&gt;<br>&gt; extensions.conf:<br>&gt; [general]<br>&gt; static=yes<br>&gt; writeprotect=no<br>&gt; autofallthrough=yes<br>&gt; clearglobalvars=no<br>&gt; priorityjumping=no<br>&gt;<br>&gt; [globals]<br>&gt; ATTENDANT=1001<br>&gt; OUTBOUNDTRUNK=ZAP/g1<br>&gt;<br>&gt; [extentions]<br>&gt; exten =&gt; _10XX,1,Ringing<br>&gt; exten =&gt; _10XX,2,Dial(SIP/${EXTEN},20)<br>&gt; exten =&gt; _10XX,3,Answer<br>&gt; exten =&gt; _10XX,4,VoiceMail(u${EXTEN}@voicemail)<br>&gt; exten =&gt; _10XX,5,Hangup<br>&gt;<br>&gt; [voicemail]<br>&gt; exten =&gt; _910XX,1,Wait(1)<br>&gt; exten =&gt; _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)<br>&gt;<br>&gt; [local]<br>&gt; include =&gt; extentions<br>&gt; include =&gt; voicemail<br>&gt;<br>&gt; [incoming]<br>&gt; exten =&gt; s,1,Answer<br>&gt; exten =&gt;
 s,2,Background(our-voicemail-sound)<br>&gt; exten =&gt; t,1,Playback(vm-goodbye)<br>&gt; exten =&gt; t,2,Hangup( )<br>&gt; exten =&gt; 0,1,Dial(SIP/${ATTENDANT},20)<br>&gt; exten =&gt; 1,1,Directory(voicemail,internal,f)<br>&gt; exten =&gt; 2,1,Directory(voicemail,internal)<br>&gt; include =&gt; extentions<br>&gt;<br>&gt; [local-trunks]<br>&gt; exten =&gt; _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})<br>&gt; exten =&gt; _9XXXXXXXXXX,2,Congestion( )<br>&gt; exten =&gt; _9XXXXXXXXXX,102,Congestion( )<br>&gt; exten =&gt; 911,1,Dial(${OUTBOUNDTRUNK}/911)<br>&gt; exten =&gt; 9911,1,Dial(${OUTBOUNDTRUNK}/911)<br>&gt;<br>&gt; [local-access]<br>&gt; ignorepat =&gt; 9<br>&gt; include =&gt; local<br>&gt; include =&gt; local-trunks<br>&gt;<br>&gt;<br>&gt; zapata.conf:<br>&gt;<br>&gt; [trunkgroups]<br>&gt; [channels]<br>&gt; context=default<br>&gt; switchtype=national<br>&gt; signalling=fxo_ls<br>&gt; rxwink=300              ; Atlas seems to use long (250ms) winks<br>&gt;
 usecallerid=yes<br>&gt; hidecallerid=no<br>&gt; callwaiting=yes<br>&gt; usecallingpres=yes<br>&gt; callwaitingcallerid=yes<br>&gt; threewaycalling=yes<br>&gt; transfer=yes<br>&gt; canpark=yes<br>&gt; cancallforward=yes<br>&gt; callreturn=yes<br>&gt; echocancel=yes<br>&gt; echocancelwhenbridged=yes<br>&gt; rxgain=0.0<br>&gt; txgain=0.0<br>&gt; group=1<br>&gt; callgroup=1<br>&gt; pickupgroup=1<br>&gt; immediate=no<br>&gt; group=1<br>&gt; echocancel=yes<br>&gt; switchtype=national<br>&gt; signalling=fxs_ks<br>&gt; context=incoming<br>&gt; echocancelwhenbridged=yes<br>&gt; channel =&gt; 1-4<br>&gt;<br>&gt;<br>&gt; /etc/zaptel.conf:<br>&gt; fxsks=1,2,3,4<br>&gt; loadzone = us<br>&gt; defaultzone=us<br>&gt;<br>&gt; log:<br>&gt; Asterisk Ready.<br>&gt;     -- Star ting simple switch on 'Zap/1-1'<br>&gt; Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18<br>&gt; (Ring Begin)...<br>&gt; Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie<br>&gt; made mylen
 &lt; 0 (-155)<br>&gt; Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID<br>&gt; feed failed: Success<br>&gt; Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID<br>&gt; returned with error on channel 'Zap/1-1'<br>&gt;     -- Executing Answer("Zap/1-1", "") in new stack<br>&gt;     -- Executing BackGround("Zap/1-1", "our-voicemail-sound") in new stack<br>&gt;     -- Playing 'our-voicemail-sound' (language 'en')<br>&gt;   == CDR updated on Zap/1-1<br>&gt;     -- Executing Ringing("Zap/1-1", "") in new stack<br>&gt;     -- Executing Dial("Zap/1-1", "SIP/1000|20") in new stack<br>&gt;     -- Called 1000<br>&gt;     -- SIP/1000-54e4 is ringing<br>&gt;     -- SIP/1000-54e4 an swered Zap/1-1<br>&gt;   == Spawn extension (incoming, 1000, 2) exited non-zero on 'Zap/1-1'<br>&gt;     -- Hungup 'Zap/1-1'<br>&gt;<br>&gt; ------------------------------------------------------------------------<br>&gt; Bring words and photos together (easily) with<br>&gt;
 PhotoMail<br>&gt; <http:><br>&gt; - it's free and works with Yahoo! Mail.<br>&gt;<br>&gt;------------------------------------------------------------------------<br>&gt;<br>&gt;_______________________________________________<br>&gt;--Bandwidth and Colocation provided by Easynews.com --<br>&gt;<br>&gt;Asterisk-Users mailing list<br>&gt;To UNSUBSCRIBE or update options visit:<br>&gt;   http://lists.digium.com/mailman/listinfo/asterisk-users<br>&gt;  <br>&gt;<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br>   http://lists.digium.com/mailman/listinfo/asterisk-users<br></http:></blockquote><br></div><p>
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