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Hello, I have been running into some issues lately using various versions of 4.x installed with the CentOS ISO. Incoming calls via my ITSP (SIP trunk) work great, both ends can hear each other and the quality is great. However, outgoing calls have one-way audio issues. I can hear them fine, but they cannot hear me. When I use sipviewer it does not show any errors that I see, but when I look at packet captures using SIP Workbench or Wireshark I do not see sipxbridge sending any audio traffic out as shown in the included screenshot. The only non-standard setting I have is that I had to change my RTP audio ports to 10000-20000 per my ITSP's recommendations and as part of that change the SIP port used by freeswitch to 25060. A little about my configuration: Using stable and dev releases of 4.x. Primary phone for testing: Polycom IP550 using 3.1.2 and 3.1.3 SIP software Secondary phone for testing: Standard landline with Linksys / Sipura SPA2102 Both phones auto-configured using sipXconfig web interface Server has a public static address with no NAT in use. Thank you, Scott  |
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