[OpenSIPS-Users] 405 from UAS to UAC

[Vlad Paiu]

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:1001 at 192.168.2.102>;tag=CCB10274-C7949905.
> To:<sip:1001 at 192.168.2.102>.
> CSeq: 1 REGISTER.
> Call-ID: 87ecdd18-8826a1a6-a85fcd57 at 192.168.2.11.
> Contact:<sip:1001 at 192.168.2.11>;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:1001 at 192.168.2.102>;tag=CCB10274-C7949905.
> To:<sip:1001 at 192.168.2.102>;tag=4899a85fdda7a45fc4d7b6eb4e737879.aa2b.
> CSeq: 1 REGISTER.
> Call-ID: 87ecdd18-8826a1a6-a85fcd57 at 192.168.2.11.
> 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','mpeer at test.com','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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