1.6.2 installed but registered deces create this error on cli

I have a softphone and a sipura ata registered

Name/username Host Dyn Nat ACL Port Status
201/201 D N A 5061 OK (13 ms)
203/203 D N A 5070 OK (5 ms)

But when I dial a extention from a extention I get this on cli

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

Any ideas? is this a bug?


I did not say in this original posting where I have obtained Freepbx but it was the iso download from Asterisknow web site. I have also found out that the gui/db is NOT writing the route-out includes under [ext-local]. Also, when dialing out, there is no dial-plan information being echoed to the cli other then rtp sound packet info, which is two lines of information. In /var/log/asterisk/full shows the dial out errors in cli. What it was saying was that the outbound rout extensions were not found in [ext-local]. My guess was that some were in the asterisk code or db code, the includes from out-route trunk context were not included. I created a temporary solution by coping the 002-out route context from extensions_additional.conf and pasted them in extensions_custom.conf file. I will see if there is a fix for this issue. In the meantime, I can use my asterisk system with this band aid approach.

Seems for some unknown reason, the gui did not pass the context changes to the extensions. Had to manually edit the conf files to reflect the changes. Both extensions can call each other but the sipura ata, for some unknown reason, is not sending a ring tone to the phone. Sure, the soft phone can call the ata, I pickup the phone, and can hear my self though the soft phone speaker. Will find out the issue with the ring generator settings of the ata.