[asterisk-users] Ringtones won't work

Mike Haney (Gmail) Sun, 15 Oct 2006 10:56:39 -0700

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the seemingly simple things do not want to work for me:

- Ringtones. Apparently the phones do not have any of the defaults on them as the Ring Type menu on each phone lists "ms, Inc." beside each option, and will not play anything. I've placed several .wav files (from http://www.voipphreak.ca/index.php?serendipity%5Baction%5D=search&serendipity%5BsearchTerm%5D=ringtones ) and set up the sip.cfg as per what I've been able to find, a copy is below. The phones do download the .wav files on each boot, and list the filenames in the web browser config pages, but still show "ms, Inc." under the Ring Type menu?

- Busy indicators/presence. I have configured a buddy watcher on the 601 which will show the appropriate Online/On Phone status of the 301 through the Buddies menu, but it does not inicate the status from the directory key/button on the main screen. Should the indicator beside the contact name not show some sort of status update when the associated buddy is on the phone?

- Voicemail. This one is just odd, and I have only found one search result that has the same issue but unfortunately no resolution. When either phone connects to voicemail they are presented with the voice prompts but any key I press is not recognized (ie. Press 1 for new messages and the voice prompts just continue like nothing was pressed). This happens through onetouch voicemail and by dialing the VM extension directly (I can't even log in if dialing the VM extension directly).

If anyone can shed some light on these topics it would be greatly appreciated!

Many thanks,
Mike

MY CURRENT SIP.CFG:
-------------------------------------------------------------------------------
<?xml version="1.0" standalone="yes"?>

<sip>
   <voIpProt>
      <local voIpProt.local.port="5060"/>
      <server voIpProt.server.1.address="10.215.100.1" voIpProt.server.1.port="" voIpProt.server.1.transport="UDPonly" voIpProt.server.1.expires="3600" voIpProt.server.1.register="1" voIpProt.server.1.retryTimeOut="0" voIpProt.server.1.retryMaxCount="0" voIpProt.server.1.expires.lineSeize="30"/>
      <SIP voIpProt.SIP.useRFC2543hold= "1" voIpProt.SIP.lcs="0" voIpProt.SIP.sendCompactHdrs="0" voIpProt.SIP.WM50="0" voIpProt.SIP.keepalive.sessionTimers="0" voIpProt.SIP.requestURI.E164.addGlobalPrefix="">
         <outboundProxy voIpProt.SIP.outboundProxy.address="" voIpProt.SIP.outboundProxy.port="5060"/>
         <alertInfo voIpProt.SIP.alertInfo.1.value="AA" voIpProt.SIP.alertInfo.1.class= "3"/>
         <alertInfo voIpProt.SIP.alertInfo.2.value="RA" voIpProt.SIP.alertInfo.2.class="4"/>
         <requestValidation voIpProt.SIP.requestValidation.1.request="" voIpProt.SIP.requestValidation.1.method="" voIpProt.SIP.requestValidation.1.request.1.event="">
            <digest voIpProt.SIP.requestValidation.digest.realm=" 10.215.100.1"/>
         </requestValidation>
         <specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard="1" voIpProt.SIP.specialEvent.checkSync.alwaysReboot="0"/>
         <conference voIpProt.SIP.conference.address=""/>
      </SIP>
   </voIpProt>
   <dialplan dialplan.impossibleMatchHandling="2" 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>
   <sampled_audio saf.1="SoundPointIPWelcome.wav" saf.2="RING_bennyhill.wav" saf.3="RING_drwho.wav" saf.4="RING_inspectorgadget.wav" saf.5="RING_jamesbond.wav" saf.6="RING_knightrider.wav" saf.7="RING_macgyver.wav" saf.8="RING_missionimpossible.wav" saf.9="RING_nightcourt.wav" saf.10="RING_ateam.wav"/>
   <HTTPD httpd.enabled= "1" httpd.cfg.enabled="1" httpd.cfg.port="80"/>
   <feature feature.1.name="presence" feature.1.enabled="1"/>
   <logging>
      <level>
         <change log.level.change.sip="4" log.level.change.sip.obs="5"/>
      </level>
   </logging>
</sip>

EXCERPT EXTENSIONS.CONF:
-------------------------------------------------------------------------------
exten => 120,hint,SIP/120
exten => 120,1,Macro(extensions,SIP/120,120)
exten => 120,2,Dial(SIP/120)
exten => 120,3,Answer
exten => 120,4,Set(TIMEOUT(response)=10)
exten => 120,5,Playback(NoAnswer_Extension)
exten => 120,6,Voicemail(u120)
exten => 120,n,Hangup

exten => 158,hint,SIP/158
exten => 158,1,Macro(extensions,SIP/158,158)
exten => 158,2,Dial(SIP/158)
exten => 158,3,Answer
exten => 158,4,Set(TIMEOUT(response)=10)
exten => 158,5,Playback(NoAnswer_Extension)
exten => 158,6,Voicemail(u158)
exten => 158,n,Hangup

EXCERPT SIP.CONF:
-------------------------------------------------------------------------------
[120]
type=friend
context=local
username=120
password=12345
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw
progressinband=no
callerid=Reception <120>

[158]
type=friend
context=local
username=158
password=12345
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw
progressinband=no
callerid=IT Department <158>

--
Mike Haney
http://mikeandkatslife.ca/

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