Polycom sound station 6000 with ip office

[mg1986]

Ok, I've frequented this forum for help and have found a lot of good advice here. This is my first time posting though.

I am running 8.1 version 67 and have purchased and installed my 3rd party license. Never used a non Avaya phone with ip office and its whipping me, My system status tells me the extension is trying to register but fails because of incorrect user id/password. I'm fairly certain they all match up though. I'm attaching a monitor reading when it tries to register. Any help would be greatly appreciated. And yea, conference is spelled wrong everywhere......

426184699mS SIP Rx: UDP 192.168.1.23:5060 -> 192.168.1.1:5060
REGISTER sip:192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKcd798cd1B96A8A36
From: "confrence" <sip:119@192.168.1.1>;tag=AAC7D622-99153763
To: <sip:119@192.168.1.1>
CSeq: 1 REGISTER
Call-ID: 94b549a7-2c793874-775f68ad@192.168.1.23
Contact: <sip:119@192.168.1.23>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.2.0917
Max-Forwards: 70
Expires: 3600
Content-Length: 0

426184703mS SIP Tx: UDP 192.168.1.1:5060 -> 192.168.1.23:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKcd798cd1B96A8A36
From: "confrence" <sip:119@192.168.1.1>;tag=AAC7D622-99153763
To: <sip:119@192.168.1.1>;tag=ca25c9fb29dabedb
Call-ID: 94b549a7-2c793874-775f68ad@192.168.1.23
CSeq: 1 REGISTER
User-Agent: IP Office 8.1 (67)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH

http://WWW-Authenticate:

Digest nonce="44664028b16ab99f1a5b",realm="ipoffice",algorithm=MD5
Supported: timer
Server: IP Office 8.1 (67)
Content-Length: 0

426184710mS SIP Rx: UDP 192.168.1.23:5060 -> 192.168.1.1:5060
REGISTER sip:192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKe222aac097DB4F49
From: "confrence" <sip:119@192.168.1.1>;tag=AAC7D622-99153763
To: <sip:119@192.168.1.1>
CSeq: 2 REGISTER
Call-ID: 94b549a7-2c793874-775f68ad@192.168.1.23
Contact: <sip:119@192.168.1.23>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.2.0917
Authorization: Digest username="", realm="ipoffice", nonce="44664028b16ab99f1a5b", uri="sip:192.168.1.1:5060", response="2822a7bb80f97d7f32f37d24894559d1", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0

426184715mS SIP Tx: UDP 192.168.1.1:5060 -> 192.168.1.23:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bKe222aac097DB4F49
From: "confrence" <sip:119@192.168.1.1>;tag=AAC7D622-99153763
To: <sip:119@192.168.1.1>;tag=67594a6b48af8e37
Call-ID: 94b549a7-2c793874-775f68ad@192.168.1.23
CSeq: 2 REGISTER
User-Agent: IP Office 8.1 (67)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH
Supported: timer
Server: IP Office 8.1 (67)

 

[amriddle01]

The username will usually be the extn number and the password is the login code not the user passsord

 

[mg1986]

Thanks for the quick response, I'll change that and hopefully be done! .... for now

 

[mg1986]

Ok, I've got it to register now (Thanks amriddle) now I can make internal calls fine (inbound and outbound) but it will not dial external numbers? Any ideas, I have no experience with polycom phones. Below is a monitor from trying to make a call, I never even see where it tries to dial the outbound number


432431801mS SIP Rx: UDP 192.168.1.23:5060 -> 192.168.1.1:5060
SUBSCRIBE sip:VM.confrence@192.168.1.1:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bK253292283ACD69C1
From: "Conference Phone" <sip:119@192.168.1.1>;tag=A7F05D62-62704E63
To: "224" <sip:224@192.168.1.1>;tag=1228b343f6c87ba1
CSeq: 7 SUBSCRIBE
Call-ID: f2f24417-cc1e942c-a59860d5@192.168.1.23
Contact: <sip:119@192.168.1.23>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: message-summary
User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.2.0917
Accept: application/simple-message-summary
Max-Forwards: 70
Expires: 3600
Content-Length: 0

432431804mS SIP Tx: UDP 192.168.1.1:5060 -> 192.168.1.23:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.23;branch=z9hG4bK253292283ACD69C1
From: "Conference Phone" <sip:119@192.168.1.1>;tag=A7F05D62-62704E63
To: "224" <sip:224@192.168.1.1>;tag=1228b343f6c87ba1
Call-ID: f2f24417-cc1e942c-a59860d5@192.168.1.23
CSeq: 7 SUBSCRIBE
Expires: 180
Contact: "224" <sip:224@192.168.1.1:5060;transport=udp>
Supported: timer,100rel
Server: IP Office 8.1 (67)
Content-Length: 0

432431806mS SIP Tx: UDP 192.168.1.1:5060 -> 192.168.1.23:5060
NOTIFY sip:119@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bKd6cbc4d27131e162b2d0c9cfa3f0ec91
From: "224" <sip:224@192.168.1.1>;tag=1228b343f6c87ba1
To: "Conference Phone" <sip:119@192.168.1.1>;tag=A7F05D62-62704E63
Call-ID: f2f24417-cc1e942c-a59860d5@192.168.1.23
CSeq: 8 NOTIFY
Contact: "confrence" <sip:VM.confrence@192.168.1.1:5060;transport=udp>
Supported: timer,100rel
Max-Forwards: 70
Event: message-summary
Subscription-State: active
Content-Type: application/simple-message-summary
Content-Length: 98

Messages-Waiting: yes
Message-Account: sip:VM.confrence@192.168.1.1
Voice-Message: 1/0 (0/0)

432431816mS SIP Rx: UDP 192.168.1.23:5060 -> 192.168.1.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;rport;branch=z9hG4bKd6cbc4d27131e162b2d0c9cfa3f0ec91
From: "224" <sip:224@192.168.1.1>;tag=1228b343f6c87ba1
To: "Conference Phone" <sip:119@192.168.1.1>;tag=A7F05D62-62704E63
CSeq: 8 NOTIFY
Call-ID: f2f24417-cc1e942c-a59860d5@192.168.1.23
Contact: <sip:119@192.168.1.23>
Event: message-summary
User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.2.0917
Content-Length: 0

 

[mg1986]

Never mind, didn't have the port in polycom