Thank you. *José Pablo Méndez *********
On Sun, Dec 18, 2011 at 8:23 PM, Kevin P. Fleming <[email protected]>wrote: > On 12/18/2011 01:22 PM, José Pablo Méndez Soto wrote: > > Embarrassingly enough, I just tried the nat=no again both in the >> general and peer sections and the blessed phone registered.... My >> apologies, again, I wrote the thread late at night probably this blinded >> me. >> > > No problem, we've all done that :-) > > Now, one question about a previous answer from you ("It is exactly that; >> 'force_rport' is now the default....."): >> >> is the trigger for using the source UDP port from the REGISTER, inside >> the rport field and inside the destination UDP port of the 200 OK: >> >> 1. The mismatch between the UDP source port of the REGISTER and the VIA >> port? Or >> 2. The fact that the other entity sends an empty rport? >> 3. Or any of the above? >> >> >> Its a difficult question to ask/describe, so if I am not asking >> correctly please let me know. Thanks a lot, really. >> > > Not at all. The trigger for Asterisk to respond to the port that the > request was sent from, instead of the port listed in the top-most Via > header, is *exactly* 'force_rport'. This causes Asterisk to behave as if > the 'rport' parameter was included in the top-most Via header, which would > be an explicit request from the sending UA for Asterisk to respond to the > sending port (and also report back what the sending port was, but that's > not part of the problem here). > > So, if the sending UA include an empty 'rport' parameter in its top-most > Via header, *or* if the Asterisk has been told to act as if one had been > included even if it wasn't, then Asterisk will respond to the perceived > sending port; otherwise, it will respond to the port listed in the top-most > Via header. > > As far as we know from our research before making this change, the Cisco > phones in question are the only ones that send their requests from one port > and require the responses to go back to a different port. All other phones > that we (and others) use with Asterisk use the same port for both, which > makes them quite easy to use behind NAT devices. The Cisco phone models you > are dealing with won't work behind a NAT device unless that NAT device has > a 'helper' that understands SIP and can fix up this situation (and of > course many Cisco phone users have Cisco routers that do exactly this). > > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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