Since it said fail to authenticate user, why not try to register without
username/password?
Kun
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Michael
Graves
Sent: Thursday, August 18, 2005 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] trouble with IP500
Hello All,
I've spent a day trying to get a Polycom IP500 wokring with my Asterisk
box. I have several others that are working fine, but this one is
getting by me. Can someone on-list tell from the following SIP debug
what I've missed?
Sip read:
INVITE sip:2000@192.168.1.30:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E
From: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
To: <sip:2000@192.168.1.30;user=phone>
CSeq: 1 INVITE
Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37
Contact: <sip:2004@192.168.1.37:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 225
v=0
o=- 1124335166 1124335166 IN IP4 192.168.1.37
s=Polycom IP Phone
c=IN IP4 192.168.1.37
t=0 0
a=sendrecv
m=audio 2224 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
14 headers, 10 lines
Using latest request as basis request
Sending to 192.168.1.37 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.37:2224
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104
(ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E
From: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
To: <sip:2000@192.168.1.30;user=phone>;tag=as2c798834
Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000@192.168.1.30>
Proxy-Authenticate: Digest realm="asterisk",
nonce="006c685d"
Content-Length: 0
to 192.168.1.37:5060
Scheduling destruction of call 'a9092ab-b63e7115-89ce2c58@192.168.1.37'
in 15000 ms
Found user '2004'
pbx*CLI>
Sip read:
ACK sip:2000@192.168.1.30:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E
From: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
To: <sip:2000@192.168.1.30;user=phone>;tag=as2c798834
CSeq: 1 ACK
Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37
Contact: <sip:2004@192.168.1.37:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Max-Forwards: 70
Content-Length: 0
11 headers, 0 lines
pbx*CLI>
Sip read:
INVITE sip:2000@192.168.1.30:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK893143acACFB06A9
From: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
To: <sip:2000@192.168.1.30;user=phone>
CSeq: 2 INVITE
Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37
Contact: <sip:2004@192.168.1.37:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="default",
realm="asterisk",
nonce="006c685d",
uri="sip:2000@192.168.1.30:5060;user=phone",
response="57abe54c660e517d81086bd4f40ad628", algor
ithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 225
v=0
o=- 1124335166 1124335166 IN IP4 192.168.1.37
s=Polycom IP Phone
c=IN IP4 192.168.1.37
t=0 0
a=sendrecv
m=audio 2224 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
15 headers, 10 lines
Using latest request as basis request
Sending to 192.168.1.37 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.37:2224
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104
(ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Found user '2004'
Aug 17 22:19:30 NOTICE[456]: chan_sip.c:7660 handle_request: Failed to
authenticate user "2004"
<sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK893143acACFB06A9
From: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
To: <sip:2000@192.168.1.30;user=phone>;tag=as2c798834
Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000@192.168.1.30>
Content-Length: 0
to 192.168.1.37:5060
pbx*CLI>
Sip read:
ACK sip:2000@192.168.1.30:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK893143acACFB06A9
From: "2004" <sip:2004@192.168.1.30>;tag=53ED9FBF-D06765E2
To: <sip:2000@192.168.1.30;user=phone>;tag=as2c798834
CSeq: 2 ACK
Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37
Contact: <sip:2004@192.168.1.37:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Max-Forwards: 70
Content-Length: 0
11 headers, 0 lines
Destroying call 'a9092ab-b63e7115-89ce2c58@192.168.1.37'
Thanks,
Michael Graves
--
Michael Graves mgraves@pixelpower.com
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. mgraves@mstvp.com
o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245
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