Hi Bogdan, Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ?
I am trying to transfer a call using the refer method as that is what the cisco phones use. The network is setup like so opensips proxy à asterisk gateway à carrier Scenario: Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone A presses xfer button and call is dropped. Any help would be appreciated. Regards, Brian Southworth From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list <firstname.lastname@example.org>; Brian Southworth <brian.southwo...@clocom.uk> Subject: Re: [OpenSIPS-Users]  WARNING:rr:after_strict: no socket found to match RR [XXX.XXX.XXX.XXX:5060] Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com <http://www.opensips-solutions.com> OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam <http://www.opensips.org/events/Summit-2018Amsterdam> On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA:  WARNING:rr:after_strict: no socket found to match RR [xxx.xxx.xxx.xxx:5060] When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please. Regards, Brian Southworth _______________________________________________ Users mailing list Users@lists.opensips.org <mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
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