[Asterisk-Users] Polycom registration

[Anon]

On Friday 23 April 2004 05:09 pm, Roger wrote:
> I have a PolyCom Soundpoint 500 sip phone.  I'm tring to get the phone
> registered on an asterisk box but am having no luck.
....
> Any ideas or docs I could look at to get this Polycom phone setup?

Man, do I feel you pain!  ;)  I had the same registration problem on my new 
IP 600 phone until just yesterday.  There seems to be a trick to get the 
Polycom SoundPoint IP phones to register.  In your sip.conf file, MAKE SURE 
you have "host=dynamic" for your IP 500.  Here is the entry for my correctly 
working IP 600 phone in my sip.conf file:

[500]
type=friend
host=dynamic
username=500
secret=MyPassword
canreinvite=no
context=Polycom
dtmfmode=rfc2833
qualify=300


My Asterisk server lives on 192.168.1.3, and my IP 600 lives on 192.168.1.10.  
Also, I attached all the relevent files for my IP 600 that live on its FTP 
server.

Something else that is required to get your phone to register are some details 
in the FTP server file sip.cfg,  It is critical to have the address of the 
Asterisk server set in 2 places in this file.  See my sip.cfg file and note 
where I have "192.168.1.3".  You need to substitute the address of your 
Asterisk server where it says "192.168.1.3".  You must also make sure the 
port is set to 5060 in both port variables in this file (see my sip.cfg).

See my phone1.cfg file.  Where it says reg.1.server.1.address="192.168.1.3", 
substitute "192.168.1.3" with the address of your Asterisk Server, and be 
sure the next variable in the file is set to reg.1.server.1.port="5060".

Make sure your user IDs and passwords match from the sip.conf file to the 
phone1.cfg files.

Just in case, also attached is my correctly working sip.conf file.

I REALLY how that helps!  Please tell me how you progress.

Anon
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;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use 
; SIP/username at domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
; 
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user at proxyhostname 
; where the proxyhostname is defined in a section below 
; 
; Useful CLI commands to check peers/users:
;   sip show peers		Show all SIP peers (including friends)
;   sip show users		Show all SIP users (including friends)
;   sip show registry		Show status of hosts we register with
;
;   sip debug			Show all SIP messages
;

[general]
port = 5060			; Port to bind to
bindaddr = 192.168.1.3		; Address to bind SIP channel to
context = INVALID		; Default context for incoming calls
;srvlookup = yes		; Enable DNS SRV lookups on outbound calls
				; Asterisk only uses the first host in SRV records
;pedantic = yes			; Enable slow, pedantic checking for Pingtel
tos=lowdelay			; IP QoS parameter, either keyword or value
				; like tos=184
;maxexpirey=3600		; Max length of incoming registration we allow
;defaultexpirey=120		; Default length of incoming/outoing registration
;notifymimetype=text/plain	; Allow overriding of mime type in NOTIFY
;videosupport=yes		; Turn on support for SIP video

disallow=all			; Disallow all codecs
allow=ulaw			; Allow codecs in order of preference
;allow=ilbc

; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
; needs to be defined in extensions.conf to be able to accept calls
; from this SIP proxy (provider)
;
; host is either a host name defined in DNS or the name of a 
; section defined below.
;
; Examples:
;
;register => 1234:password at mysipprovider.com	
;
;    Will call to the 's' extension
;
;
;register => 2345 at mysipprovider.com/1234 	
;
;    Register 2345 at sip provider.  Calls from this provider connect to local 
;    extension 1234 in extensions.conf default context, unless you define 
;    [mysipprovider.com] in a section below, and configure a context


;externip = 200.201.202.203	; Address that we're going to put in outbound SIP messages
				; if we're behind a NAT
;localnet = 192.168.1.0		; Internal NETWORK address
;localmask = 255.255.255.0     	; Internal netmask
				; The externip, localnet and localmask is used
				; when registering and communication with other proxies
				; that we're registered with


;========================== Polycom 1 ========================================================
[500]
type=friend
host=dynamic
;defaultip=192.168.1.10
username=500
secret=5555
canreinvite=no
context=Polycom
dtmfmode=rfc2833
;mailbox=
qualify=300
;nat=


;========================== End Polycom 1 ====================================================
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