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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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Vincent (C) 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> -------------------
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