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 <[email protected]> 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. > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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