On Fri, May 29, 2009 at 1:02 PM, Scott <[email protected]> wrote: > 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.
sipxrelay is the component that relays packets (not sipxbridge). sipxbridge/relay has been tested against systems that use Cisco gear. Of course there are many variables that can all conspire to cause you problems. I assume you changed the NAT port range using sipxconfig. If your relay is behind a NAT you need to check "Server behind NAT" in the NAT Traversal config screen. It is possible that there is something bad with the SDP that is being sent to sipxbridge during call setup that is making it impossible for sipxrelay to forward the packets. I would need to look at the signaling to make sure. You would need to provide more information (such as a sipx-snapshot ) to make a determination. Can you post etc/sipxpbx/nattraversalrules.xml? I'd need to see a detailed trace for further analysis. Please mail me a sipx-snapshot output. Please set sipxbridge to DEBUG logging level and INFO on sipxproxy and sipx registrar. Stop the system. Clean out the /var/log/sipxpbx directory of all old logs. Restart the system and run the problematic call flow. I would also like to look at a wireshark trace captured during call setup of the problematic call illustrating the problem. Thank you. Ranga > > 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 > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > -- M. Ranganathan _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
