I've just installed 1.8.3-rc3 on a test server as we really needed that deadlock involving REFER fix on our server but now I'm having an odd issue with one way audio with a specific type of call.
If I do extension to extension calls there is full 2 way audio. If I route in an incoming call through numbers provided by our SIP provider there is no inbound audio (mobile to * SIP extension) but there is outbound audio (* SIP extension to mobile). If I route a call through our production server (1.4.17 debian) to a second identity on the same SIP phone as the previous condition there is perfect 2 way audio. I did suspect it might have been the firewall on the test server but I did the same call with the firewall turned off and still only had one way audio. Has anyone else experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _____________________________________________________________________ -- 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
