[Asterisk-Users] Re: transfers/parked calls + polycom 501

sdgesa gaeharth pollux1234567890 at yahoo.com
Thu Mar 16 12:56:44 MST 2006 

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.
  
  <sip>
    .......
     <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1">
        <digitmap  dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT"  dialplan.digitmap.timeOut="3"/>
        <routing>
           <server  dialplan.routing.server.1.address=""  dialplan.routing.server.1.port="5060"/>
           <emergency  dialplan.routing.emergency.1.value="911"  dialplan.routing.emergency.1.server.1="1"/>
        </routing>
     </dialplan>
  .....
  </sip>

  
  Here is extensions.conf as well:
  
  [general]
  static=yes
  writeprotect=no
  autofallthrough=yes
  clearglobalvars=no
  priorityjumping=no
  
  [globals]
  ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009
  OUTBOUNDTRUNK=ZAP/g1
  
  [meetme-ext]
  exten => 600,1,MeetMe(1234|Mp|98765)
  
  [extentions]
  include => parkedcalls
  include => meetme-ext
  include => direct-to-voicemail
  exten => _10XX,1,Dial(SIP/${EXTEN},20,tT)
  exten => _10XX,n,Answer
  exten => _10XX,n,VoiceMail(u${EXTEN}@voicemail)
  exten => _10XX,n,Hangup()
  
  [voicemail]
  exten => _910XX,1,Wait(1)
  exten => _910XX,n,VoiceMailMain(${EXTEN:1}@voicemail)
  
  [direct-to-voicemail]
  exten => _810XX,1,VoiceMail(u${EXTEN:1}@voicemail)
  exten => _810XX,n,Hangup()
  
  [local]
  include => extentions
  include => voicemail
  
  [incoming]
  exten => s,1,Answer
  exten => s,n,Wait(2)
  exten => s,n,Set(TIMEOUT(response)=15)
  exten => s,n,Background(intro)
  exten => s,n,WaitExten()
  exten => s,n,Playback(vm-goodbye)
  exten => s,n,Hangup()
  exten => 0,1,Dial(${ATTENDANT},20,tT)
  exten => 0,n,Playback(vm-nobodyavail)
  exten => 0,n,Hangup()
  exten => 1,1,Directory(voicemail,extentions,f)
  exten => 2,1,Directory(voicemail,extentions)
  include => meetme-ext
  include => extentions
  exten => i,1,Playback(pbx-invalid)
  exten => i,2,Goto(incoming,s,1)
  exten => t,1,Playback(vm-goodbye)
  exten => t,2,Hangup()
  
  [outbound]
  ignorepat => 9
  include => parkedcalls
  exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,tT)
  exten => _9XXXXXXXXXX,2,Congestion()
  exten => _9XXXXXXXXXX,102,Congestion()
  exten => _91900NXXXXXX,1,Congestion()
  exten => _91976NXXXXXX,1,Congestion()
  exten => _91[123456789]XXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,tT)
  exten => _91[123456789]XXNXXXXXX,2,Congestion()
  exten => _91[123456789]XXNXXXXXX,102,Congestion()
  exten => 9911,1,Dial(${OUTBOUNDTRUNK}/ww911)
  exten => 9411,1,Dial(${OUTBOUNDTRUNK}/ww411)
  exten => 0,1,Dial(${OUTBOUNDTRUNK}/ww0)
  
  [local-access]
  include => local
  include => outbound
  
  thanks
  
  
Sean Cook <scook at kinex.net> wrote:  -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

No... did you get this from the sip.cfg or did you assume that the
default is there?  Asterisk will send a 404 back to the phone if the
entry does not exist.... but if it is just sending before you are
finished then there is a problem... what do you have the TimeOut set for?

Sean

sdgesa gaeharth wrote:
> [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT
> 
> I have never had this changed on any phones. This should be the default.
> 
> Does this value change based on what extensions are available to the
> phone via asterisk extensions file? In other words, does asterisk tell
> the phone what extensions are available and then the polycoms change the
> map themselves?
> 
> thanks
> 
> 
> 
> 
> */Sean Cook /* wrote:
> 
> This sounds like a digitmap issue... from your sip.cfg what is your
> digitmap set to?
> 
> Sean
> 
> sdgesa gaeharth wrote:
>> I am using the latest firmware and bootrom and this is a problem with
>> all 12 polycom 501s that we have in the office. If I want to transfer
>> to 1005 for example while on the p hone with the original caller,
> I press
>> transfer -> blind -> type "1", "0" then the phone clears the display
>> and the transfer fails. It only allows me to dial the first two digits
>> of the extension I want to transfer to. It even happens when I dial
>> local sip to local sip, not just sip to pstn. This seems like a config
>> mistake I made.....
> 
>> thanks
> 
> 
>> */Noah Miller /* wrote:
> 
>> Hi -
> 
>> > I am not sure what I did but blind transfers do not work. The
>> Polycom does
>> > not allow me to dial the extension of the person I want to
>> transfer to after
>> > I hit:
>> >
>> > transfer -> blind
> 
>> I would strongly suggest getting the latest firmware, and using the
>> sample
>> configuration files with that firmware to set up your phone. Th is
> SHOULD
>> work. If it still does not work after doing this, there may be a
>> hardware
>> issue with your phone.
> 
>> - Noah
> 
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