[OpenSIPS-Users] 405 from UAS to UAC

[Sammy Govind]

Hey Nik,

Its not OpenSIPS sending REGISTER to Polycom but Responding Polycom that
the method(register) it is request isn't allowed - this is coming from your
configuration.

If you just want a load-balancer on top of asterisks have you followed the
opensips wiki page for integrating asterisk real-time with opensips, Im
sure you do.

trans=200 could also be calls=200 as I've seen it running like this as
well. Its just resource group name and max allowed resource count.

Regards,
Sammy.

On Sun, Nov 13, 2011 at 5:27 AM, Nick Khamis <symack at gmail.com> 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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