Darren McIntosh
2003-Dec-01 03:52 UTC
[Asterisk-Users] Re: Asterisk behind NAT << How to do it. (Leif Madsen)
> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.comIn 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. darren
Leif Madsen
2003-Dec-01 11:38 UTC
[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 <leif@hacklocalhost.com> http://www.hacklocalhost.com
David J Carter
2003-Dec-08 12:46 UTC
[Asterisk-Users] Re: Asterisk behind NAT << How to do it.(Leif Madsen)
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: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Leif Madsen Sent: 01 December 2003 18:39 To: asterisk-users@lists.digium.com 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 hostaddress> 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 <leif@hacklocalhost.com> http://www.hacklocalhost.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
William Waites
2003-Dec-08 13:12 UTC
[Asterisk-Users] Re: Asterisk behind NAT << How to do it.(Leif Madsen)
On Mon, Dec 08, 2003 at 07:46:50PM -0000, David J Carter wrote:> Hi, > > I have chan_sip.c version 1.259 do I still need the patch.yes. -- /~\ The ASCII Ribbon Campaign \ / No HTML/RTF in email X No Word docs in email / \ Respect for open standards
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