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Jachin Rupe

2005-Sep-27 10:21 UTC

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

hi there

I'm setting up asterisk@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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Dean Collins

2005-Sep-27 10:57 UTC

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

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

 

Cheers,

Dean

 

________________________________

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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 asterisk@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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Matt Gibson

2005-Sep-27 11:01 UTC

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

Jachin Rupe wrote:> hi there
>
> I'm setting up asterisk@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.
>Sounds like it might be your DTMF handling.

Try setting relaxdtmf=yes in your sip.conf,
and if running cvs head, change

dtmfmode=auto for sip peers

if not using HEAD, then try

dtmfmode=rfc2833 (g729 codec)
or
dtmfmode=inband (ulaw/alaw)


do your debug logs show anything like double digits, or missing digits 
when you try calling IVR's?

matt

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