RE: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers

Tue, 27 Sep 2005 11:05:08 -0700

Change your dtmf setting. Covered lots of times before, or info on voip-info.com

 

Cheers,

Dean

 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jachin Rupe
Sent: Tuesday, 27 September 2005 1:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom IP 500 - problem dialing extra numbers

 

hi there

 

I'm setting up [EMAIL PROTECTED] 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

_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to