In sip.cfg located in my ftp for the phones, I see what is below. It looks to be the same as what I see when I log into the http server on each phone.<br> <br> <sip><br> .......<br> <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"><br> <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/><br> <routing><br> <server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="5060"/><br> <emergency dialplan.routing.emergency.1.value="911" dialplan.routing.emergency.1.server.1="1"/><br> </routing><br> </dialplan><br> .....<br> </sip><br><br> <br> Here is extensions.conf as well:<br>
<br> [general]<br> static=yes<br> writeprotect=no<br> autofallthrough=yes<br> clearglobalvars=no<br> priorityjumping=no<br> <br> [globals]<br> ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009<br> OUTBOUNDTRUNK=ZAP/g1<br> <br> [meetme-ext]<br> exten => 600,1,MeetMe(1234|Mp|98765)<br> <br> [extentions]<br> include => parkedcalls<br> include => meetme-ext<br> include => direct-to-voicemail<br> exten => _10XX,1,Dial(SIP/${EXTEN},20,tT)<br> exten => _10XX,n,Answer<br> exten => _10XX,n,VoiceMail(u${EXTEN}@voicemail)<br> exten => _10XX,n,Hangup()<br> <br> [voicemail]<br> exten => _910XX,1,Wait(1)<br> exten => _910XX,n,VoiceMailMain(${EXTEN:1}@voicemail)<br> <br> [direct-to-voicemail]<br> exten => _810XX,1,VoiceMail(u${EXTEN:1}@voicemail)<br> exten => _810XX,n,Hangup()<br> <br> [local]<br> include => extentions<br> include => voicemail<br> <br> [incoming]<br> exten => s,1,Answer<br> exten =>
s,n,Wait(2)<br> exten => s,n,Set(TIMEOUT(response)=15)<br> exten => s,n,Background(intro)<br> exten => s,n,WaitExten()<br> exten => s,n,Playback(vm-goodbye)<br> exten => s,n,Hangup()<br> exten => 0,1,Dial(${ATTENDANT},20,tT)<br> exten => 0,n,Playback(vm-nobodyavail)<br> exten => 0,n,Hangup()<br> exten => 1,1,Directory(voicemail,extentions,f)<br> exten => 2,1,Directory(voicemail,extentions)<br> include => meetme-ext<br> include => extentions<br> exten => i,1,Playback(pbx-invalid)<br> exten => i,2,Goto(incoming,s,1)<br> exten => t,1,Playback(vm-goodbye)<br> exten => t,2,Hangup()<br> <br> [outbound]<br> ignorepat => 9<br> include => parkedcalls<br> exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,tT)<br> exten => _9XXXXXXXXXX,2,Congestion()<br> exten => _9XXXXXXXXXX,102,Congestion()<br> exten => _91900NXXXXXX,1,Congestion()<br> exten => _91976NXXXXXX,1,Congestion()<br> exten
=> _91[123456789]XXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,tT)<br> exten => _91[123456789]XXNXXXXXX,2,Congestion()<br> exten => _91[123456789]XXNXXXXXX,102,Congestion()<br> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)<br> exten => 9411,1,Dial(${OUTBOUNDTRUNK}/ww411)<br> exten => 0,1,Dial(${OUTBOUNDTRUNK}/ww0)<br> <br> [local-access]<br> include => local<br> include => outbound<br> <br> thanks<br> <br> <br><b><i>Sean Cook <scook@kinex.net></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> -----BEGIN PGP SIGNED MESSAGE-----<br>Hash: SHA1<br><br>No... did you get this from the sip.cfg or did you assume that the<br>default is there? Asterisk will send a 404 back to the phone if the<br>entry does not exist.... but if it is just sending before you are<br>finished then there is a problem... what do you have the TimeOut set for?<br><br>Sean<br><br>sdgesa
gaeharth wrote:<br>> [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT<br>> <br>> I have never had this changed on any phones. This should be the default.<br>> <br>> Does this value change based on what extensions are available to the<br>> phone via asterisk extensions file? In other words, does asterisk tell<br>> the phone what extensions are available and then the polycoms change the<br>> map themselves?<br>> <br>> thanks<br>> <br>> <br>> <br>> <br>> */Sean Cook <scook @kinex.net="">/* wrote:<br>> <br>> This sounds like a digitmap issue... from your sip.cfg what is your<br>> digitmap set to?<br>> <br>> Sean<br>> <br>> sdgesa gaeharth wrote:<br>>> I am using the latest firmware and bootrom and this is a problem with<br>>> all 12 polycom 501s that we have in the office. If I want to transfer<br>>> to 1005 for example while on the p hone with the original caller,<br>> I
press<br>>> transfer -> blind -> type "1", "0" then the phone clears the display<br>>> and the transfer fails. It only allows me to dial the first two digits<br>>> of the extension I want to transfer to. It even happens when I dial<br>>> local sip to local sip, not just sip to pstn. This seems like a config<br>>> mistake I made.....<br>> <br>>> thanks<br>> <br>> <br>>> */Noah Miller /* wrote:<br>> <br>>> Hi -<br>> <br>>> > I am not sure what I did but blind transfers do not work. The<br>>> Polycom does<br>>> > not allow me to dial the extension of the person I want to<br>>> transfer to after<br>>> > I hit:<br>>> ><br>>> > transfer -> blind<br>> <br>>> I would strongly suggest getting the latest firmware, and using the<br>>> sample<br>>> configuration files with that firmware to set up your phone. Th is<br>> SHOULD<br>>> work.
If it still does not work after doing this, there may be a<br>>> hardware<br>>> issue with your phone.<br>> <br>>> - Noah<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>>> Brings words and photos together (easily) with<br>>> PhotoMail<br>> <br>>> - it's free and works with Yahoo! Mail.<br>> <br>> <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>- --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>> Yahoo! Mail<br>> Use Photomail<br>> <http: //pa.yahoo.com/*http://us.rd.yahoo.com/evt="38867/*http://photomail.mail.yahoo.com"><br>> to share photos without annoying attachments.<br><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>-----BEGIN PGP SIGNATURE-----<br>Version: GnuPG v1.4.2 (MingW32)<br>Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org<br><br>iD8DBQFEGb+zy9wPyZpnL2URAjktAKCXDE7C2K/sLIdMFz8jfUAIU1oDDACfa1hQ<br>RtE7sUkbqSOWZZBjFTT4Czo=<br>=UqXr<br>-----END PGP SIGNATURE-----<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:></scook></blockquote><br><p>
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