[Asterisk-Users] Polycom IP 500 - problem dialing extra numbers
Jachin Rupe
jachin at voltz.net
Tue Sep 27 10:21:44 MST 2005
hi there
I'm setting up asterisk at home and I'm using Polycom IP 500 phones.
When I call a number that has a digital receptionist (i.e. "dial 1 or
such and such, dial 2 for this and that...") the Polycom doesn't seem
to send the extra digits. When I try it with X-Lite things appear to
work fine, so I think the problem is with the Polycom configuration.
Here's some of my configuration files. If I didn't included an
important one please let me know.
---------
000000000000.cfg
---------
<?xml version="1.0" standalone="yes"?>
<!-- Default Master SIP Configuration File-->
<!-- Edit and rename this file to <Ethernet-address>.cfg for each
phone.-->
<!-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -->
<APPLICATION APP_FILE_PATH="sip.ld"
CONFIG_FILES="phone1.cfg, sip.cfg"
MISC_FILES=""
LOG_FILE_DIRECTORY="/log/" />
---------
sip.cfg
---------
<?xml version="1.0" standalone="yes"?>
<!-- SIP Application Configuration File -->
<!-- $Revision: 1.63 $ $Date: 2004/11/08 18:52:16 $ -->
<sip>
<voIpProt>
<local voIpProt.local.port=""/>
<server voIpProt.server.1.address="10.0.20.0"
voIpProt.server.1.port="5060"
voIpProt.server.1.transport="DNSnaptr"
voIpProt.server.1.expires="300"
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="10.0.20.0"
voIpProt.SIP.outboundProxy.port="5060" />
<alertInfo voIpProt.SIP.alertInfo.1.value=""
voIpProt.SIP.alertInfo.1.class="" />
<requestValidation voIpProt.SIP.requestValidation.
1.request=""
voIpProt.SIP.requestValidation.
1.method=""
voIpProt.SIP.requestValidation.
1.request.1.event="">
<digest
voIpProt.SIP.requestValidation.digest.realm="PolycomSPIP" />
</requestValidation>
<specialEvent
voIpProt.SIP.specialEvent.lineSeize.nonStandard="1"
voIpProt.SIP.specialEvent.checkSync.alwaysReboot="0"/>
<conference voIpProt.SIP.conference.address="" />
</SIP>
</voIpProt>
<dialplan dialplan.impossibleMatchHandling="0"
dialplan.removeEndOfDial="1">
<digitmap dialplan.digitmap="" 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>
<logging>
<level>
<change log.level.change.sip="4"
log.level.change.sip.obs="5"/>
</level>
</logging>
</sip>
------------
I just realized something... I don't have a phone1.cfg file, should I?
I adopted this system in a partial working state from someone else
and I'm still figuring out why things are the way they are.
thanks
-jachin
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