Brian Vincent (C)
2006-Jun-21 10:09 UTC
[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@coppercolorado.com
______________________________________________________________________________________________________________
Confidentiality Warning: This message and any attachments are intended only for
the use of the intended recipient(s),
are confidential, and may be privileged. If you are not the intended recipient,
you are hereby notified that any review,
retransmission, conversion to hard copy, copying, circulation or other use of
this message and any attachments is strictly
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immediately by return e-mail, and delete this
message and any attachments from your system. Thank you.
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Jerry Jones
2006-Jun-21 11:31 UTC
[Asterisk-Users] Polycom 601 problems with multiple registrations
Couple thoughts/observations Yes you WILL get this working:) since your .cfg is just a snippet and not the complete lines, kinda hard to know exactly. Hopefully you have the complete file and are just sending the relevant pieces to the list. Also, if you enable sip debug, do you even see registration messages from the other lines? On Jun 21, 2006, at 12:09 PM, Brian Vincent (C) wrote:> 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@coppercolorado.com > > > > > > ______________________________________________________________________ > ________________________________________ > > Confidentiality Warning: This message and any attachments are > intended only for the use of the intended recipient(s), > are confidential, and may be privileged. If you are not the > intended recipient, you are hereby notified that any review, > retransmission, conversion to hard copy, copying, circulation or > other use of this message and any attachments is strictly > prohibited. If you are not the intended recipient, please notify > the sender immediately by return e-mail, and delete this > message and any attachments from your system. Thank you. > ______________________________________________________________________ > ________________________________________ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Brian Vincent (C)
2006-Jun-21 12:25 UTC
[Asterisk-Users] Polycom 601 problems with multiple registrations
> Yes you WILL get this working:)Thanks for the encouragement.. I need it right now. The install just popped up and we're scrambling to put the pieces together from another project in the works.> since your .cfg is just a snippet and not the complete lines, kinda > hard to know exactly.Nope - that's all I'm using. Barebones config just to see what I need to get multiple registrations. Got any example cfg files where this is working that I could try out?> Also, if you enable sip debug, do you even see registration messages > from the other lines?Good tip. Well, I only see the first line registering. Here's everything from sip debug, nothing here seems to point to the problem: <-- SIP read from 10.20.0.20:5060: REGISTER sip:10.20.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.20:5060;branch=z9hG4bK4cf9f95f817CE2A4 From: "21" <sip:21@10.20.0.1>;tag=B69CC11-216A1550 To: <sip:21@10.20.0.1> CSeq: 1 REGISTER Call-ID: 3bc84d15-a5a0897b-a84d6b02@10.20.0.20 Contact: <sip:21@10.20.0.20:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Max-Forwards: 70 Expires: 3600 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.20.0.20 : 5060 (non-NAT) Transmitting (no NAT) to 10.20.0.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.20.0.20:5060;branch=z9hG4bK4cf9f95f817CE2A4;received=10.20.0.20 From: "21" <sip:21@10.20.0.1>;tag=B69CC11-216A1550 To: <sip:21@10.20.0.1> Call-ID: 3bc84d15-a5a0897b-a84d6b02@10.20.0.20 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:21@10.20.0.1> Content-Length: 0 --- Transmitting (no NAT) to 10.20.0.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.20.0.20:5060;branch=z9hG4bK4cf9f95f817CE2A4;received=10.20.0.20 From: "21" <sip:21@10.20.0.1>;tag=B69CC11-216A1550 To: <sip:21@10.20.0.1>;tag=as4da0e355 Call-ID: 3bc84d15-a5a0897b-a84d6b02@10.20.0.20 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:21@10.20.0.1> WWW-Authenticate: Digest realm="asterisk", nonce="796a1aa2" Content-Length: 0 --- Scheduling destruction of call '3bc84d15-a5a0897b-a84d6b02@10.20.0.20' in 15000 ms co990228*CLI> <-- SIP read from 10.20.0.20:5060: REGISTER sip:10.20.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.20:5060;branch=z9hG4bK5eba690e23AEDE57 From: "21" <sip:21@10.20.0.1>;tag=B69CC11-216A1550 To: <sip:21@10.20.0.1> CSeq: 2 REGISTER Call-ID: 3bc84d15-a5a0897b-a84d6b02@10.20.0.20 Contact: <sip:21@10.20.0.20:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Authorization: Digest username="21", realm="asterisk", nonce="796a1aa2", uri="sip:10.20.0.1", response="b2c501a01084c61122e6710e60100365", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 10.20.0.20 : 5060 (non-NAT) Transmitting (no NAT) to 10.20.0.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.20.0.20:5060;branch=z9hG4bK5eba690e23AEDE57;received=10.20.0.20 From: "21" <sip:21@10.20.0.1>;tag=B69CC11-216A1550 To: <sip:21@10.20.0.1> Call-ID: 3bc84d15-a5a0897b-a84d6b02@10.20.0.20 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:21@10.20.0.1> Content-Length: 0 --- Transmitting (no NAT) to 10.20.0.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.20.0.20:5060;branch=z9hG4bK5eba690e23AEDE57;received=10.20.0.20 From: "21" <sip:21@10.20.0.1>;tag=B69CC11-216A1550 To: <sip:21@10.20.0.1>;tag=as4da0e355 Call-ID: 3bc84d15-a5a0897b-a84d6b02@10.20.0.20 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: <sip:21@10.20.0.20:5060>;expires=3600 Date: Wed, 21 Jun 2006 19:22:14 GMT Content-Length: 0 --- Scheduling destruction of call '3bc84d15-a5a0897b-a84d6b02@10.20.0.20' in 15000 ms 12 headers, 3 lines Reliably Transmitting (no NAT) to 10.20.0.20:5060: NOTIFY sip:21@10.20.0.20:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.1:5060;branch=z9hG4bK3586b37b From: "Unknown" <sip:Unknown@10.20.0.1>;tag=as2aa6a174 To: <sip:21@10.20.0.20:5060> Contact: <sip:Unknown@10.20.0.1> Call-ID: 436be570457e305b7ab1689421d5ed69@10.20.0.1 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:asterisk@10.20.0.1 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '436be570457e305b7ab1689421d5ed69@10.20.0.1' in 15000 ms co990228*CLI> <-- SIP read from 10.20.0.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.20.0.1:5060;branch=z9hG4bK3586b37b From: "Unknown" <sip:Unknown@10.20.0.1>;tag=as2aa6a174 To: <sip:21@10.20.0.20:5060>;tag=FE7195CD-7791E9BC CSeq: 102 NOTIFY Call-ID: 436be570457e305b7ab1689421d5ed69@10.20.0.1 Contact: <sip:21@10.20.0.20:5060> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '436be570457e305b7ab1689421d5ed69@10.20.0.1' Destroying call '3bc84d15-a5a0897b-a84d6b02@10.20.0.20' ------------------- Brian Vincent Copper Mountain Telecom vincentb@coppercolorado.com ______________________________________________________________________________________________________________ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. ______________________________________________________________________________________________________________
Bill Gibbs
2006-Jun-21 13:32 UTC
[Asterisk-Users] Polycom 601 problems with multiple registrations
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian
Vincent (C)
Sent: Wednesday, June 21, 2006 1:10 PM
To: asterisk-users@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@coppercolorado.com
________________________________________________________________________
______________________________________
Confidentiality Warning: This message and any attachments are intended
only for the use of the intended recipient(s),
are confidential, and may be privileged. If you are not the intended
recipient, you are hereby notified that any review,
retransmission, conversion to hard copy, copying, circulation or other
use of this message and any attachments is strictly
prohibited. If you are not the intended recipient, please notify the
sender immediately by return e-mail, and delete this
message and any attachments from your system. Thank you.
________________________________________________________________________
______________________________________
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Brian Vincent (C)
2006-Jun-21 16:43 UTC
[Asterisk-Users] Polycom 601 problems with multiple registrations
> Good tip. Well, I only see the first line registering. Here's > everything from sip debug, nothing here seems to point to the problem:For completeness, I'll include my solution this problem: use a different phone. For some reason the 601 I was testing with only allowed 1 registration. Maybe something in the admin setup on the phone? I'll probably do a factory reset on the phone to fix it. Anyway, I guess I found a new and strange way to break a 601. I'm not sure whether to be excited I found the solution or pissed at myself for taking so long to try a different phone. Thanks to Jerry for the tips. ------------------- Brian Vincent Copper Mountain Telecom vincentb@coppercolorado.com ______________________________________________________________________________________________________________ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. ______________________________________________________________________________________________________________