On Thu, 13 Jan 2011 09:43:26 -0500, Bruce B <bruceb...@gmail.com> wrote: >In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also >make sure you have your externip setup as well. Else you will notice one way >audio or cut off after 30 seconds.
I don't have sip_nat.conf, as I don't use any GUI to configure Asterisk. I didn't have to change anything to Asterisk as compared to when connecting directly. Since the other extensions live in the same LAN as Asterisk, should I configure "localnet" just for the remote extension that connects in through OpenVPN, while leaving 192.168.0.0/24 for the local extensions? The only issue I notice, is that Asterisk doesn't tell the other end when the local end has hung up, so the other end either remains online or hangs up after 20-30 seconds. I've tried XLite and ZoIPer, same result. This never happens when not going through the VPN. Has someone seen this? Here's the error message: ============ -- Executing [siemens@internal:1] Dial("SIP/remote-00d22b1c", "SIP/siemens") in new stack -- Called siemens -- SIP/siemens-00d329ec is ringing -- SIP/siemens-00d329ec answered SIP/remote-00d22b1c -- Packet2Packet bridging SIP/remote-00d22b1c and SIP/siemens-00d329ec == Spawn extension (internal,siemens, 1) exited non-zero on 'SIP/remote-00d22b1c' WARNING[82]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission NWQ2NTRhMzYxZjIzZTBhODY3NTBhYzMxMTk5MTUyYjY. for seqno 2 (Critical Response) ============ Thank you. -- _____________________________________________________________________ -- 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