The one way audio issues discussed recently were when an asterisk gateway with a PRI card (sangoma) were installed in asterisk and using that as a gateway. Not an ideal solution in my opinion. The problem went away if the asterisk solution was backrevved from 1.6 to 1.4.
"sipxbridge uses re-INVITE extensively (including session timer and media renegotiation). If re-INVITE does not work a lot of things are going to break." The question is do you have a call trace? On Mon, Dec 7, 2009 at 2:09 PM, [email protected] < [email protected]> wrote: > I have sipXecs 4.0.4 running on a small deployment, all with Polycom phones > (650 & 670). External calls are routed via a SIP ITSP (vitelity.net). > Recently, I've noticed that external calls lasting 30 minutes or more will > become one way audio right at the 30 minute mark. I don't recall this > happening in 4.0.2, but not many call do not last 30 minutes. > > Other than this one issue, everything else seems to be working correctly. > > All phones are on the same subnet as the sipXecs server (from the ISO). > The LAN is NATed with a pfSense router (port randomization disabled for > sipXecs). > > I noticed some emails recently about calls disconnecting after 30 minutes > as well. Could these be related to the issue I am experiencing? > > Any suggestions on how to resolve the one-way audio issue? > > Thanks! > Robert > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
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