[Asterisk-Users] Polycom IP500

Tim Jackson tim at angelinacounty.net
Thu Jan 6 07:54:01 MST 2005 

They were updated, to reflect the new card. And I can call in perfectly.


-Tim

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 2:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500

How about your zapata.conf and zaptel.conf files?  Were they updated for
the new card?

W

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Polycom IP500

Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone
cards to a 1U IBM server with a TDM04B card. I finally got the card
working in the server, but I'm having issues with these Polycom IP500s
now. Using the exact same config from the old server I'm getting weird
errors. Dial a number on the phone and it gives you dialtone but no user
interaction (if that makes sense) then after about 35-40 seconds it
displays "Line used remotely" and hangs up. Inbound calls ring, but you
can't answer them, registration seems to be ok, but I'm at a loss.

sip.conf:
[101]
type=friend
callerid="Tim Jackson - Home" <101>
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all

OR

[101]
type=peer
callerid="Tim Jackson" <101>
secret=itsasekret
host=dynamic
dtmfmode=rfc2833
nat=yes
mailbox=101
canreinvite=no
context=default
disallow=all
allow=ulaw


app log from phone:

0106005724|res  |4|00|[ResFinderC]: Failed to download file BigLogo.bmp,
errno 0xdd.
0106005724|net  |*|00|Initial log entry. Current logging level 4 key  
0106005724||*|00|Initial log entry. Current logging level 4 ssps 
0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2
FS1, Version=1.1.2.0002 17-May-04 15:28
0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112.
0106005724|pps  |*|00|Initial log entry. Current logging level 4 sip  
0106005724||*|00|Initial log entry. Current logging level 4 cfg  
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg  
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|cfg  |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|cfg  |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|so   |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101]
0106005727|cfg  |5|00|Edit|Simple|error 0x0 when locking (setting
lcl.ml.lang="")
0106005727|cfg  |5|00|Edit|Simple|error 0x0 when locking (param get) 
0106005727|slog |*|00|Initial log entry. Current logging level 4
0106005927|cfg  |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz


debug sip peer 101

Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=36767043-B9FDB2DA
To: <sip:101 at 192.9.200.9>
CSeq: 1 REGISTER
Call-ID: d8038d0f-22c84c59-3f42a480 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Max-Forwards: 70
Expires: 3600
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 192.9.202.2 : 5060 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=36767043-B9FDB2DA
To: <sip:101 at 192.9.200.9>;tag=as024fe72d
Call-ID: d8038d0f-22c84c59-3f42a480 at 192.9.202.2
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 192.9.200.9>
Content-Length: 0


 to 192.9.202.2:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=36767043-B9FDB2DA
To: <sip:101 at 192.9.200.9>;tag=as024fe72d
Call-ID: d8038d0f-22c84c59-3f42a480 at 192.9.202.2
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 192.9.200.9>
WWW-Authenticate: Digest realm="angelinacounty.net", nonce="243b35d1"
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480 at 192.9.202.2'
in 15000 ms
asterisk*CLI>

Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=36767043-B9FDB2DA
To: <sip:101 at 192.9.200.9>
CSeq: 2 REGISTER
Call-ID: d8038d0f-22c84c59-3f42a480 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Authorization: Digest username="101", realm="angelinacounty.net",
nonce="243b35d1", uri="sip:192.9.200.9:5060",
response="11f3478d812d35993018150f29fb5e81", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.9.202.2 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=36767043-B9FDB2DA
To: <sip:101 at 192.9.200.9>;tag=as024fe72d
Call-ID: d8038d0f-22c84c59-3f42a480 at 192.9.202.2
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 192.9.200.9>
Content-Length: 0


 to 192.9.202.2:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=36767043-B9FDB2DA
To: <sip:101 at 192.9.200.9>;tag=as024fe72d
Call-ID: d8038d0f-22c84c59-3f42a480 at 192.9.202.2
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:101 at 192.9.202.2:5060>;expires=3600
Date: Thu, 06 Jan 2005 06:46:36 GMT
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call 'd8038d0f-22c84c59-3f42a480 at 192.9.202.2'
in 15000 ms
asterisk*CLI>

Sip read:
SUBSCRIBE sip:100 at 192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=A5DE6FC-D938162B
To: <sip:100 at 192.9.200.9>
CSeq: 1 SUBSCRIBE
Call-ID: 70bce7a8-79a1e882-74df3bc1 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Max-Forwards: 70
Expires: 3600
Content-Length: 0


13 headers, 0 lines
Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 :
5060 (non-NAT) Found peer '101'
Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKfaf3aa6eF088ADF7;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=A5DE6FC-D938162B
To: <sip:100 at 192.9.200.9>;tag=as77cf03d0
Call-ID: 70bce7a8-79a1e882-74df3bc1 at 192.9.202.2
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:100 at 192.9.200.9>
Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7d0b7e8a"
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1 at 192.9.202.2'
in 15000 ms
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:101 at 192.9.202.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport
From: "asterisk" <sip:asterisk at 192.9.200.9>;tag=as00270f99
To: <sip:101 at 192.9.202.2:5060>
Contact: <sip:asterisk at 192.9.200.9>
Call-ID: 23c9fa48037fec98416d74650481661e at 192.9.200.9
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 42

Messages-Waiting: no
Voice-Message: 0/0
 (NAT) to 192.9.202.2:5060
Scheduling destruction of call
'23c9fa48037fec98416d74650481661e at 192.9.200.9' in 15000 ms asterisk*CLI>

Sip read:
SUBSCRIBE sip:100 at 192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=A5DE6FC-D938162B
To: <sip:100 at 192.9.200.9>
CSeq: 2 SUBSCRIBE
Call-ID: 70bce7a8-79a1e882-74df3bc1 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Proxy-Authorization: Digest username="101", realm="angelinacounty.net",
nonce="7d0b7e8a", uri="sip:100 at 192.9.200.9:5060",
response="38d9b121d4ee361e584727823f195810", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


14 headers, 0 lines
Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 :
5060 (NAT) Found peer '101'
Looking for 100 in default
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bK25cf07353B4FBD16;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=A5DE6FC-D938162B
To: <sip:100 at 192.9.200.9>;tag=as5a181df1
Call-ID: 70bce7a8-79a1e882-74df3bc1 at 192.9.202.2
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:100 at 192.9.200.9>;expires=3600
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call '70bce7a8-79a1e882-74df3bc1 at 192.9.202.2'
in 3610000 ms
Reliably Transmitting:
NOTIFY sip:101 at 192.9.200.9 SIP/2.0
Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport
From: <sip:100 at 192.9.200.9>;tag=as5a181df1
To: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=A5DE6FC-D938162B
Contact: <sip:100 at 192.9.200.9>
Call-ID: 70bce7a8-79a1e882-74df3bc1 at 192.9.202.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Content-Type: application/xpidf+xml
Content-Length: 339

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN"
"xpidf.dtd">
<presence>
<presentity uri="sip:101 at 192.9.200.9;method=SUBSCRIBE" /> <atom
id="100"> <address uri="sip:100 at 192.9.200.9;user=ip"
priority="0,800000"> <status status="open" /> <msnsubstatus
substatus="online" /> </address> </atom> </presence>
 (NAT) to 192.9.202.2:5060
asterisk*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK4b1c9378;rport
From: "asterisk" <sip:asterisk at 192.9.200.9>;tag=as00270f99
To: <sip:101 at 192.9.202.2:5060>;tag=81B3E4D0-500C78DF
CSeq: 102 NOTIFY
Call-ID: 23c9fa48037fec98416d74650481661e at 192.9.200.9
Contact: <sip:101 at 192.9.202.2:5060>
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Content-Length: 0


10 headers, 0 lines
Destroying call '23c9fa48037fec98416d74650481661e at 192.9.200.9'
asterisk*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK51df898e;rport
From: <sip:100 at 192.9.200.9>;tag=as5a181df1
To: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=A5DE6FC-D938162B
CSeq: 102 NOTIFY
Call-ID: 70bce7a8-79a1e882-74df3bc1 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Content-Length: 0


9 headers, 0 lines
Message is NOTIFY
asterisk*CLI>

Sip read:
SUBSCRIBE sip:101 at 192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
To: <sip:101 at 192.9.200.9>
CSeq: 1 SUBSCRIBE
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Accept: application/simple-message-summary
Max-Forwards: 70
Expires: 3600
Content-Length: 0


14 headers, 0 lines
Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 :
5060 (non-NAT) Found peer '101'
Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bK244f75296C7E782A;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
To: <sip:101 at 192.9.200.9>;tag=as1d96eff1
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 192.9.200.9>
Proxy-Authenticate: Digest realm="angelinacounty.net", nonce="7735e16f"
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call '3c57a613-cc4f559d-1ed53124 at 192.9.202.2'
in 15000 ms
asterisk*CLI>

Sip read:
SUBSCRIBE sip:101 at 192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
To: <sip:101 at 192.9.200.9>
CSeq: 2 SUBSCRIBE
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Accept: application/simple-message-summary
Proxy-Authorization: Digest username="101", realm="angelinacounty.net",
nonce="7735e16f", uri="sip:101 at 192.9.200.9:5060",
response="c5f05dd1a6463189b10e6217b2c61f48", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0


15 headers, 0 lines
Using latest SUBSCRIBE request as basis request Sending to 192.9.202.2 :
5060 (NAT) Found peer '101'
Looking for 101 in default
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
To: <sip:101 at 192.9.200.9>;tag=as2d05161a
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:101 at 192.9.200.9>
Content-Length: 0


 to 192.9.202.2:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.9.202.2:5060;branch=z9hG4bKc9f21bf8C5677891;received=192.9.202.2;rpo
rt=5060
From: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
To: <sip:101 at 192.9.200.9>;tag=as2d05161a
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:101 at 192.9.200.9>;expires=3600
Content-Length: 0


 to 192.9.202.2:5060
Scheduling destruction of call '3c57a613-cc4f559d-1ed53124 at 192.9.202.2'
in 3610000 ms
Reliably Transmitting:
NOTIFY sip:101 at 192.9.200.9 SIP/2.0
Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport
From: <sip:101 at 192.9.200.9>;tag=as2d05161a
To: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
Contact: <sip:101 at 192.9.200.9>
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 201

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="full" entity="sip:101 at 192.9.200.9"> <dialog id="101">
<state>confirmed</state> </dialog> </dialog-info>
 (NAT) to 192.9.202.2:5060
asterisk*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.9.200.9:5060;branch=z9hG4bK7a9eec3d;rport
From: <sip:101 at 192.9.200.9>;tag=as2d05161a
To: "Tim Jackson" <sip:101 at 192.9.200.9>;tag=B0756DC7-622969BE
CSeq: 102 NOTIFY
Call-ID: 3c57a613-cc4f559d-1ed53124 at 192.9.202.2
Contact: <sip:101 at 192.9.202.2:5060>
Event: dialog
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Content-Length: 0


10 headers, 0 lines
Message is NOTIFY
Destroying call 'd8038d0f-22c84c59-3f42a480 at 192.9.202.2'



Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile

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