[Asterisk-Users] Polycom 601 problems with multiple registrations

Wed Jun 21 13:32:38 MST 2006

This does work.  I have a few phones with 1.5.something doing this.

I didn't fill out the reg.x.server.x.address field - so it uses the
sip.cfg default.

Here's a snippet of what worked on a 601 - 6 line keys a few days ago:

reg.1.displayName="x110" 

reg.1.address="110" 

reg.1.label="x110" 

reg.1.type="private" 

reg.1.thirdPartyName="" 

reg.1.auth.userId="110" 

reg.1.auth.password="DURRR" 

reg.1.server.1.address="" 

reg.1.server.1.port="" 

reg.1.server.1.transport="DNSnaptr" 

reg.1.server.2.transport="DNSnaptr" 

reg.1.server.1.expires="120" 

reg.1.server.1.register="1" 

reg.1.server.1.retryTimeOut="" 

reg.1.server.1.retryMaxCount="" 

reg.1.server.1.expires.lineSeize="" 

reg.1.acd-login-logout="0" 

reg.1.acd-agent-available="0" 

reg.1.ringType="2" 

reg.1.lineKeys="6" 

reg.1.callsPerLineKey="1"

If you want multiple registrations, just change the 110 and password to
whatever the other extension is.

Does your asterisk console show the registration?

Bill

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian
Vincent (C)
Sent: Wednesday, June 21, 2006 1:10 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Polycom 601 problems with multiple
registrations

I'm stumped on this one and any help would be greatly appreciated.

I'm just trying to get my Polycom 601 to have multiple extensions on it.
For example, on line 1 I want extension 21, on line 2 I want extension
22, and on line 3 I want extension 23.  Ideally I'd actually have each
extension appear on 2 lines and therefore filling up all 6.  I should be
able to do that with the reg.x.lineKeys parameter.  Anyway, I'm not even
at the point of getting multiple registrations to work, so I'll worry
about that later.  Right now the only thing that works is registering
the first extension - it registers just fine and works as expected.  No
matter what extension I put on there it works, but I only have line 1
working.  What am I doing wrong?

Okay, now my config.  I've got a REALLY basic set up.  I copied the
files off the wiki from krisk.org.  I completely removed ipmid.cfg
temporarily so it wouldn't interfere with this (putting it back in place
has no effect).  That leaves me with just sip.cfg and the phone cfg
file.  I'm booting with FTP.  I know the config files are loading
correctly because I can make changes and they do have an effect.  Here's
the phone20.cfg file for the phone:

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>

<!-- Example Per-phone Configuration File -->

<!-- $Revision: 1.59 $  $Date: 2004/05/22 00:50:41 $ -->

<phone1>

<reg

  reg.1.address="21"

  reg.1.auth.userId="21"

  reg.1.auth.password="21"

  reg.1.server.1.address="10.20.0.1"

  reg.2.address="22"

  reg.2.auth.userId="22"

  reg.2.auth.password="22"

  reg.2.server.1.address="10.20.0.1"

  reg.3.address="23"

  reg.3.auth.userId="23"

  reg.3.auth.password="23"

  reg.3.server.1.address="10.20.0.1" />

</phone1>

And sip.cfg:

<!-- IP Application Configuration File -->

<!--

 $Revision: 1.57.2.1 $  $Date: 2004/07/27 00:23:34 $

-->

<sip>

<voIpProt>

<local voIpProt.local.port="5060"/>

<server voIpProt.server.1.address="10.20.0.1"
voIpProt.server.1.port="5060" voIp

Prot.server.1.transport="UDPonly" voIpProt.server.1.expires="3600"
voIpProt.serv

er.1.register="1" voIpProt.server.1.retryTimeOut="0"
voIpProt.server.1.retryMaxC

ount="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="10.20.0.1"/>

</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]xxxxxx

xxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/>

        <routing>

<server dialplan.routing.server.1.address=""
dialplan.routing.server.1.port="506

0"/>

<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>

-------------------
Brian Vincent
Copper Mountain Telecom
vincentb at coppercolorado.com 

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