Hi Jeff, theoretically yes, because you have all the needed information (hmm...maybe except the NAT status from request, but you can store it via transaction)....Practically, the save() function does expect to receive a request only, so it must be changed to work with a reply also.
Regards, Bogdan Jeff Pyle wrote: > I've thought a lot about this as well, although I haven't taken it nearly as > far as John has. > > A thought: is it possible to do a save() in the reply route, only upon a > 200 OK from the end registrar? > > > - Jeff > > > > On 5/12/09 5:09 AM, "Bogdan-Andrei Iancu" <[email protected]> wrote: > > >> Hi John, >> >> This mid-registrar approach may work but it is not 100% correct as >> OpenSIPS (as mid-registrar) does not obey the actions of the final >> registrar (Asterisk). Ex: >> - Asterisk may forbid the registration and you already saved the >> registration on OpenSIPS >> - Asterisk may change the Expire time while to saved the >> registration with the expire sent by client. >> >> Anyhow, ignoring this aspects, lets go further :) : >> >> 1) is the registration scenario working ok? if not what is the exact >> problem (some trace will help). >> >> I will wait for you answer before moving further with the calling stuff. >> >> Regards, >> Bogdan >> >> John Morris wrote: >> >>> After several days of playing with OpenSIPS 1.5.0 and RTPProxy 1.2.0, I >>> have a partially working SIP+RTP ALG configuration, and have gotten stuck. >>> I could use some general advice from the list. >>> >>> The company has an Asterisk/FreePBX server on an internal network, and the >>> CEO wants to use a SIP phone from outside. Because the sip alg iptables >>> module isn't working, and in preparation for another project, I started >>> investigating OpenSIPS for use as a border proxy to connect phones across >>> NAT (and, the next project, to route a SIP trunk over a VPN from the >>> network of a DSL+phone company that intermittently blocks SIP traffic in >>> hopes of plugging revenue leaks). >>> >>> The network looks like this: >>> >>> SIP UA <-> home NAT gateway <-> Internet <-> OpenSIPS server/NAT router >>> <-> Asterisk >>> >>> The standard opensips.cfg file doesn't work as is. The SIP phone needs to >>> register to the Asterisk server directly. In addition, it seems there is >>> extra logic needed to support multiple network interfaces (mhomed=1 only >>> partially solves the problem). >>> >>> The way I've gone with this in testing is to relay REGISTERs to Asterisk, >>> but after a save("location","0x02") to enable a lookup("location") on >>> messages originating from the PBX. The phone is configured with an >>> outbound proxy, and all packets to the proxy matching "uri==myself" are >>> thrown away. This worked great on the single-interfaced, internal test >>> installation. Now that there are multiple interfaces involved, things are >>> breaking again; ACKs and BYEs are sent out the wrong interface, and >>> RTPProxy is behaving strangely in bridged mode. >>> >>> There seem to be no good configuration examples for either multi-homed >>> proxies or for proxies that relay REGISTERs. This makes me think that I'm >>> going about this the wrong way. >>> >>> Also, I have looked at other software, like siproxd, opensbc and uh, that >>> other b2bua that functions as an SBC, but none of those seem to allow this >>> REGISTER pass-through function. >>> >>> What is the best approach for this scenario? The above approach of >>> relaying REGISTERs to Asterisk? Is there maybe another approach where >>> phones register to OpenSIPS directly, and OpenSIPS in turn somehow sends >>> another REGISTER to Asterisk? Or am I missing the idea completely? >>> >>> I'd appreciate general pointers about how to proceed. I've been putting >>> some Asterisk and FreePBX tutorials and CentOS RPMs on >>> http://www.zultron.com, mostly aimed at small office-like environments. >>> Looking through various lists, this seems a highly sought-after >>> configuration. If I succeed, I'll document it in hopes of filling the gap >>> in this sort of example. >>> >>> Thanks >>> >>> John >>> >>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
