Re: [asterisk-users] No Audio with SIP to only one provider whenswitching servers

2007-04-27 Thread Hadar Pedhazur
Brad Sumrall wrote: I would not rule your firewall out as the problem! Port 5060 is only the authentication port, the rtp stream is normally 10,000 thru 20,000. Some of your phone may have STUN modules on them. Open 10,000 thru 20,000 and 5060 on the firewall. Stick some holes in it for testing

RE: [asterisk-users] No Audio with SIP to only one provider whenswitching servers

2007-04-25 Thread Brad Sumrall
I would not rule your firewall out as the problem! Port 5060 is only the authentication port, the rtp stream is normally 10,000 thru 20,000. Some of your phone may have STUN modules on them. Open 10,000 thru 20,000 and 5060 on the firewall. Stick some holes in it for testing purposes. Verify ports