<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. The outid line rings, the person picks p but no sounds.<br> <br> Any suggestions????<br> <br> thanks<br><br><b><i>Ken D'Ambrosio <ken@jots.org></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> >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>> please help!!!<br>><br>> I am dialing into our asterisk server(TDM400p) from the psnt. I hear<br>> our voicemail message and I press the extention 1000. The Polycom ip<br>> phone in the office rings. I pickup but neither side can hear one<br>>
another. What have I done wrong?<br>><br>> thanks<br>><br>> sip.conf:<br>> [general]<br>> context=local-access ; Default context for incoming calls<br>> bindport=5060 ; UDP Port to bind to (SIP standard<br>> port is 5060)<br>> bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds<br>> to all)<br>> srvlookup=yes ; Enable DNS SRV lookup s on outbound<br>> calls<br>> musicclass=default<br>><br>> [authentication]<br>><br>> [1000]<br>> username=1000<br>> regexten=1000<br>> mailbox=1000@voicemail<br>> callerid="jon Smith" <1000><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" <1001><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 => _10XX,1,Ringing<br>> exten => _10XX,2,Dial(SIP/${EXTEN},20)<br>> exten => _10XX,3,Answer<br>> exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail)<br>> exten => _10XX,5,Hangup<br>><br>> [voicemail]<br>> exten => _910XX,1,Wait(1)<br>> exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail)<br>><br>> [local]<br>> include => extentions<br>> include => voicemail<br>><br>> [incoming]<br>> exten => s,1,Answer<br>> exten =>
s,2,Background(our-voicemail-sound)<br>> exten => t,1,Playback(vm-goodbye)<br>> exten => t,2,Hangup( )<br>> exten => 0,1,Dial(SIP/${ATTENDANT},20)<br>> exten => 1,1,Directory(voicemail,internal,f)<br>> exten => 2,1,Directory(voicemail,internal)<br>> include => extentions<br>><br>> [local-trunks]<br>> exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})<br>> exten => _9XXXXXXXXXX,2,Congestion( )<br>> exten => _9XXXXXXXXXX,102,Congestion( )<br>> exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)<br>> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)<br>><br>> [local-access]<br>> ignorepat => 9<br>> include => local<br>> include => 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 ; 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 => 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>> -- Star ting simple switch on 'Zap/1-1'<br>> Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18<br>> (Ring Begin)...<br>> Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie<br>> made mylen
< 0 (-155)<br>> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID<br>> feed failed: Success<br>> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID<br>> returned with error on channel 'Zap/1-1'<br>> -- Executing Answer("Zap/1-1", "") in new stack<br>> -- Executing BackGround("Zap/1-1", "our-voicemail-sound") in new stack<br>> -- Playing 'our-voicemail-sound' (language 'en')<br>> == CDR updated on Zap/1-1<br>> -- Executing Ringing("Zap/1-1", "") in new stack<br>> -- Executing Dial("Zap/1-1", "SIP/1000|20") in new stack<br>> -- Called 1000<br>> -- SIP/1000-54e4 is ringing<br>> -- SIP/1000-54e4 an swered Zap/1-1<br>> == Spawn extension (incoming, 1000, 2) exited non-zero on 'Zap/1-1'<br>> -- Hungup 'Zap/1-1'<br>><br>> ------------------------------------------------------------------------<br>> Bring words and photos together (easily) with<br>>
PhotoMail<br>> <http:><br>> - it's free and works with Yahoo! Mail.<br>><br>>------------------------------------------------------------------------<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>> <br>><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>
<hr size=1> <BR>
What are the most popular cars? Find out at <a href="http://us.rd.yahoo.com/evt=38382/_ylc=X3oDMTEzNWFva2Y2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDMmF1dG9z/*http://autos.yahoo.com/newcars/popular/thisweek.html
">Yahoo! Autos</a>