10 mar 2013 kl. 03:04 skrev Nick Khamis <sym...@gmail.com>: > Hello Everyone, > > I have gone through a few really good tutorials from the OpenSIPS > site, Asterisk resources etc.. The unanswered question (and final > piece of our puzzle) is if it's possible to have a register free > environment in an OpenSIPS/Asterisk integration. Most approaches have > OpenSIPS relay the UA's REGISTER request to Asterisk which has > "host=dynamic" set for the Friend/Peer and everything works as > expected. > There are a lot of models for this. Check my presentation from Astricon 2010 to get some ideas. http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations
/O > Where I run into problems is in Inbound calls. When I try to call the > extension from a DID I am receiving "Unable to create channel of type > 'SIP' (cause 20 - Unknown)". And rightfully so! > Reason being: > > SIP Show Peers Yields: > > Name/username Host Dyn Forcerport ACL Port > Status Realtime > 1001/1001 192.168.2.5 N 5060 > UNREACHABLE Cached RT > TTrunk/sip.exp.com 192.168.2.5 N 5060 UNKNOWN Cached RT > > > As for who will keep track of the UA location, the OpenSIPS `location` > table has the correct > info: > > select username,domain,contact,socket from location; > +----------+--------------------+----------------------------+----------------------+ > | username | domain | contact | socket > | > +----------+--------------------+----------------------------+----------------------+ > | 1001 | sip.exp.com | sip:1001@192.168.2.11:5060 | udp:192.168.2.5:5060 | > +----------+--------------------+----------------------------+----------------------+ > > OpenSIPS: sip.exp.com > OpenSIPS: 192.168.2.5 > Asterisk: 192.168.2.10 > UA: 192.168.2.11 > > I have set `host=sip.exp.com' for the UA but the UA is still > `UNREACHABLE` by asterisk > > As for the rest of the media related stuff, everything works > perfectly. Outbound works fine. As you know, this only poses a problem > with inbound calls to the UAs. > > Your Help is Greatly Appreciated, > > Nick. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users