[asterisk-users] Ringtones won't work

[Mike Haney (Gmail)]

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 Application Configuration File -->
<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
mailbox=120 at default
disallow=all
allow=ulaw
progressinband=no
callerid=Reception <120>

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




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