Remote SIP extension deployment
Could any of you share your experience with remote SIP extensions behind another NAT? Basically my setup whould be like this: [ Remote SIP extensions]------>(private IP@) NAT Router B---(public IP @B)-------(Internet Cloud)-----------(public IP @A) NAT Router A---(private IP @A)------(private IP@C)[Freeswitch] I have the domain registered as "sip.mydomain.com" let's see, I am trying to config the remote SIP extension as 1018 using port 5080 hitting the public context, but I got the error on the freeswitch console "user 1018@sip.mydomain.com not found" Could you guys guide me to the right direction for remote SIP extensions' deploymen? Your helps are appreciate. Thanks, Chris
The double nat scenario is going to be a fun one to solve. You're better off putting FreeSWITCH on a public IP or fully understanding what is going on. The outbound profile isn't for inbound registrations by default. Its almost what you need but you'll need to turn auth-calls=true and you might need to force the registration domain. Do you have phones behind that nat with FreeSWITCH registering to port 5060? <param name="force-register-domain" value="cluecon.com"/> <-- force the domain on a sip profile. /b On Apr 27, 2008, at 10:25 AM, Chris Chen wrote:
Could any of you share your experience with remote SIP extensions behind another NAT?
Basically my setup whould be like this:
[ Remote SIP extensions]------>(private IP@) NAT Router B---(public IP @B)-------(Internet Cloud)-----------(public IP @A) NAT Router A---(private IP @A)------(private IP@C)[Freeswitch]
I have the domain registered as "sip.mydomain.com"
let's see, I am trying to config the remote SIP extension as 1018 using port 5080 hitting the public context, but I got the error on the freeswitch console
"user 1018@sip.mydomain.com not found"
Could you guys guide me to the right direction for remote SIP extensions' deploymen?
Your helps are appreciate. Thanks,
Chris
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I don't have the luck to put the freeswitch on a public IP, and I need the phones behind NAT registering with default context on port 5060 while in the meantime remote phones registered as remote SIP extensions to my freeswitch on port 5080 by public context. In the public context, I will have the outbound and inbound SIP trunks to my service providers as well. I will try your solution to see if it fits my needs. Thanks, Chris On Sun, Apr 27, 2008 at 11:32 AM, Brian West <brian@freeswitch.org> wrote:
The double nat scenario is going to be a fun one to solve. You're better off putting FreeSWITCH on a public IP or fully understanding what is going on. The outbound profile isn't for inbound registrations by default. Its almost what you need but you'll need to turn auth-calls=true and you might need to force the registration domain. Do you have phones behind that nat with FreeSWITCH registering to port 5060?
<param name="force-register-domain" value="cluecon.com"/> <-- force the domain on a sip profile.
/b
On Apr 27, 2008, at 10:25 AM, Chris Chen wrote:
Could any of you share your experience with remote SIP extensions behind another NAT?
Basically my setup would be like this:
[ Remote SIP extensions]------>(private IP@) NAT Router B---(public IP @B)-------(Internet Cloud)-----------(public IP @A) NAT Router A---(private IP @A)------(private IP@C)[Freeswitch]
I have the domain registered as "sip.mydomain.com"
let's see, I am trying to config the remote SIP extension as 1018 using port 5080 hitting the public context, but I got the error on the freeswitch console
"user 1018@sip.mydomain.com not found"
Could you guys guide me to the right direction for remote SIP extensions' deployment?
Your helps are appreciate. Thanks,
Chris
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Chris, You don't want registered users going inbound to the public context, by design this is where people that aren't authenticated end up. I fear you're close to opening up your entire dialplan to everyone which could result in toll fraud. The context the user hits is set by the user_context variable on the user no the context variable on the profile in this case. I would recommend running a copy of the default profile on port 5062 with the addition of ext-sip-ip and ext-rtp-ip and force-register-domain set to the domain in your directory you want to push all registrations on this profile thru. The profile on port 5080 will NOT authenticate calls or users and is by design for outbound service provider interaction ONLY. Personally I would use a different domain for internal vs external. /b On Apr 27, 2008, at 4:08 PM, Chris Chen wrote:
I don't have the luck to put the freeswitch on a public IP, and I need the phones behind NAT registering with default context on port 5060 while in the meantime remote phones registered as remote SIP extensions to my freeswitch on port 5080 by public context. In the public context, I will have the outbound and inbound SIP trunks to my service providers as well.
I will try your solution to see if it fits my needs. Thanks, Chris
participants (2)
-
Brian West
-
Chris Chen