Hi, I have chan_sip.c version 1.259 do I still need the patch.
I can now get calls from sipphone.com but they drop after 5 seconds. Regards Dave -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: 01 December 2003 18:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Asterisk behind NAT << How to do it.(Leif Madsen) On Mon, 2003-12-01 at 05:52, Darren McIntosh wrote: > In my configuration I have internal SIP clients registering from > 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address > of the * box as the inside_net variable the audio from 192.168.0.0/28 was > sent to the outside_addr variable giving one-way speech. Setting > internal_net to the subnet address of 192.168.0.0 and inside_mask to > 255.255.255.0 the call behaved correctly. Aha! I had not tried this configuration. Now I see how that makes more sense! I will make note of that :) Thanks Darren! -- Leif Madsen <[EMAIL PROTECTED]> http://www.hacklocalhost.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
