Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Steven Ward
Thanks. I created the gateway file in conf/directory/default/ On Mon, Mar 16, 2009 at 1:24 PM, Michael Collins wrote: > 2009/3/16 Steven Ward : > > Yes, the obvious is the case. :) I don't want to do a STUN lookup - the > two > > machines are on the same LAN. > > > > What's the best way to g

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Steven Ward
Heh heh. Guess it pays not to rush. :) Got it working now - without registering. But another thing - what if I want to set my two boxes up for registering? I see that I can set my register parameter to true, but how do I control the register string that's sent to the other box? 2009/3/16 Bria

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Brian West
First off since its not in the user directory anymore you'll have to unwrap the gateway from inside the user tags ;) /b On Mar 16, 2009, at 12:51 PM, Steven Ward wrote: Sure thing. Here it is:

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Steven Ward
Sure thing. Here it is: In vars.conf I supplied the variables' values: 2009/3/16 Brian West > I would almost bet your xml is wrong when you moved it.. care to

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Michael Collins
2009/3/16 Steven Ward : > I simply moved the file defining the gateway to conf/sip_profiles/internal > > Well, when calling from extension 1000 to 70904, what I see on the console > (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute()

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Brian West
I would almost bet your xml is wrong when you moved it.. care to share that little bit of info? /b On Mar 16, 2009, at 12:39 PM, Steven Ward wrote: I simply moved the file defining the gateway to conf/sip_profiles/ internal Well, when calling from extension 1000 to 70904, what I see on the

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Steven Ward
I simply moved the file defining the gateway to conf/sip_profiles/internal Well, when calling from extension 1000 to 70904, what I see on the console (debug mode) is: 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1...@pbx-sip-3.usa.in

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Mathieu Rene
The reason its using stun is because your external-sip-ip and external- rtp-ip params are starting with stun: As Michael says, the external profile is meant to do nat-traversal, if you dont need it, use the internal one. Math On 16-Mar-09, at 1:24 PM, Michael Collins wrote: > 2009/3/16 Steve

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Michael Collins
2009/3/16 Steven Ward : > Yes, the obvious is the case.  :) I don't want to do a STUN lookup - the two > machines are on the same LAN. > > What's the best way to get the gateway to not do a STUN lookup?  Do I need > to disable STUN for the external > profile or make this gateway use a different pro

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Steven Ward
Yes, the obvious is the case. :) I don't want to do a STUN lookup - the two machines are on the same LAN. What's the best way to get the gateway to not do a STUN lookup? Do I need to disable STUN for the external profile or make this gateway use a different profile? Thanks. SW On Mon, Mar 16,

Re: [Freeswitch-users] sip trunking question

2009-03-16 Thread Michael Collins
2009/3/16 Steven Ward : > I'm trying to set up a sip trunk between a FS and * box, and right now I'm > having trouble getting things set up so I make a call from a sip phone > registered with my FS box to a sip phone registered w/ my Asterisk box. > > I have a gateway defined as in directory/defaul

[Freeswitch-users] sip trunking question

2009-03-16 Thread Steven Ward
I'm trying to set up a sip trunk between a FS and * box, and right now I'm having trouble getting things set up so I make a call from a sip phone registered with my FS box to a sip phone registered w/ my Asterisk box. I have a gateway defined as in directory/default/example.com.xml and in my dialp