I attempted to upgrade from 1.2.10 to 1.4.1. I have 2 sip phones. The asterisk server and one of the sip phones (polycom) is inside a firewall. Another sip phone (grandstream) is outside the firewall in another firewall. We have this sip phone using a stun server.
Under 1.2.10, using the polycom phone, I can call the grandstream phone, in the asterisk console it shows ringing, I hear a ringing sound on the polycom phone. When the other person picks up the phone (grandstream), we can hear each other. The grandstream phone call dial and call the polycom phone, when the polycom user picks up the phone, they can hear the grandstream phone’s user.
After upgrading to 1.4.1, no changes were made to the firewall, asterisk config files, either phones. RTP ports were not change between versions. We just removed 1.2.10 and compiled 1.4.1 and put this asterisk version in place of 1.2.10. When the polycom user calls the grandstream phone(asterisk console shows it is the first extensions calls, but does not show the 2nd extension ringing). The polycom phone does not hear the ringing sound in the headset. Also when the grandstream phone answers, the polycom phone does not hear the grandstream phone’s voice. When the grandstream user calls the polycom phone, both users can hear each other’s voice.
Has anyone else seen this behavior between these versions? We need to upgrade to 1.4.1 for the fax machine support, unless this can be accomplished using 1.2.?? version.
How can I fix these issues?