Hello,
From what I see the Polycom ( .11 ) is sending a Register requests to
OpenSIPS ( .102 ), and most probably your are rejecting the message from
your script, with a '405 Method Not Allowed' .
So please look more carefully into your script to see what you are doing
wrong, and if you can't figure it out return and post it here.
Regards,
Vlad Paiu
OpenSIPS Developer
On 11/13/2011 02:27 AM, Nick Khamis wrote:
Hello Everyone,
I am having a hard time registering a Polycom IP301:
* 192.168.2.11 is Poly
* 192.168.2.102 is OpenSIPS
* 192.168.2.103 is Asterisk
* 192.168.2.104 is Asterisk
The following is my ngrep:
U 192.168.2.11:5060 -> 192.168.2.102:5060
REGISTER sip:192.168.2.102:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcf2ffccaEAFFA1E9.
From: "Mike Peer"<sip:[email protected]>;tag=CCB10274-C7949905.
To:<sip:[email protected]>.
CSeq: 1 REGISTER.
Call-ID: [email protected].
Contact:<sip:[email protected]>;methods="INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER".
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098.
Max-Forwards: 70.
Expires: 3600.
Content-Length: 0.
U 192.168.2.102:5060 -> 192.168.2.11:5060
SIP/2.0 405 Method Not Allowed.
Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcf2ffccaEAFFA1E9.
From: "Mike Peer"<sip:[email protected]>;tag=CCB10274-C7949905.
To:<sip:[email protected]>;tag=4899a85fdda7a45fc4d7b6eb4e737879.aa2b.
CSeq: 1 REGISTER.
Call-ID: [email protected].
Server: OpenSIPS (1.7.0-notls (i386/linux)).
Content-Length: 0.
Not quite sure why OpenSIPS is sending a REGISTER to the phone. I
know! Wrong configuration? ;)
The idea is to put a load balancing proxy, that is also in-charge of
REGSITER, between the asterisk
boxes and the clients. The entries I have in databaes are:
insert into domain value(0,'test.com',now());
insert into subscriber
values(0,'1001','astcluster.test.com','pass','[email protected]','pass','pass',null);
insert into load_balancer
values(0,1,'sip:192.168.2.103','transc=200',0,'Asterisk One');
insert into load_balancer
values(0,2,'sip:192.168.2.104','transc=200',0,'Asterisk Two');
A sligehtly off topic, I am under the impression that "transc=200",
tells our BEAUTIFUL sip proxy
that only 200 SIP calls will be sent to the media servers?
The configuration file is mostly default. I could post it if requred
Thanks in Advance,
Nick.
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