I have a site-to-site hardware (Cisco ASA-series on both ends) VPN between my house and our headquarters so I can occasionally telecommute and deal with personal calls in the office or work calls from home.
I have an Asterisk install at home [11.6.0-rc2] with a SIP connection to our HQ Asterisk box [11.7.0] and a unified dialplan between the two sites (we have a remote office location with similar hardware and software configurations, also talking to the HQ site and part of the same dialplan that has working without issue). This has been working perfectly, but at some point in the past couple weeks I can't call from home to the office - calling from the office to home still works perfectly, as do calls to/from my landline. If I call the office voicemail extension and watch the Asterisk console in SSH, it's answering the call and playing back the prompts but as far as the house is concerned, the call is never answered and after several seconds the SIP trunk is "circuit-busy" and auto fallthrough as "CONGESTION" As far as I'm aware we aren't doing any traffic shaping/filtering/blocking on either end of the VPN. From home I'm able to SSH to LAN IP of the office * box, and from the office I'm a able to SSH to the LAN IP of my home * box. Running Packet Tracer on either ASA says that it's passing traffic on port 5060 in both directions . All of the Googling I've done turns up results relating to to one-way audio -- but that's not my problem here -- if the office calls me at home, I have no problems with audio quality in either direction. I'm just not able to complete a call from home to the office. Can anyone point me in the direction of what I need to poke at with a sharp stick? Thanks in advance! Lincoln -- Lincoln King-Cliby, CTS, DMC-E-4K/T/D Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Gold) ControlWorks Consulting, LLC Crestron Services Provider | Biamp Audia Certified | Extron Control Professional -- _____________________________________________________________________ -- 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
