Problems with initial setup - basic nat
We're setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. I've wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. I've been through the wiki and reviewed list archives but I'm missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewall's external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where I'm stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won't accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. I'm sure it's something obvious but what am I missing? Thanks, Jay
Sorry to bug you guys. I figured it out. In case anyone else is just learning to crawl with freeswitch.. I enabled the following in the sip_profiles to get around the authorization errors (for now): <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/> <!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/> Then I started receiving a 404 route not found so I modified the public dialplan with the following: <extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension> Then I wasn't getting 2-way audio so I changed the sip profile for nat (which I'm using internally) and set the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the sip-ip (since I'm only using for internal nat through firewall to sip provider): <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isn't secure but it's enough to get things going. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat We're setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. I've wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. I've been through the wiki and reviewed list archives but I'm missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewall's external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where I'm stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won't accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. I'm sure it's something obvious but what am I missing? Thanks, Jay
You could have just turned auth-calls to false and context to default and accomplished the same thing ;) /b On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch….
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension>
Then I wasn’t getting 2-way audio so I changed the sip profile for nat (which I’m using internally) and set the ext-sip-ip and the ext- rtp-ip to the same value as the rtp-ip and the sip-ip (since I’m only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isn’t secure but it’s enough to get things going.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat
We’re setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. I’ve wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
I’ve been through the wiki and reviewed list archives but I’m missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewall’s external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where I’m stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won’t accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
I’m sure it’s something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@freeswitch.org
Thanks! :-) I did have auth-calls set to false in nat.xml but it wasn't working. Is there some other place I should have set this? What's the difference/application/use of the sample "public" context versus the "default" one? The sample nat.xml uses the public context. Thanks, Jay _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat You could have just turned auth-calls to false and context to default and accomplished the same thing ;) /b On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote: Sorry to bug you guys. I figured it out. In case anyone else is just learning to crawl with freeswitch. I enabled the following in the sip_profiles to get around the authorization errors (for now): <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/> <!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/> Then I started receiving a 404 route not found so I modified the public dialplan with the following: <extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension> Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider): <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. Ive been through the wiki and reviewed list archives but Im missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. Im sure its something obvious but what am I missing? Thanks, Jay _______________________________________________ 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 Brian West sip:brian@freeswitch.org
Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations. /b On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasn’t working. Is there some other place I should have set this?
What’s the difference/application/use of the sample “public” context versus the “default” one? The sample nat.xml uses the public context.
Thanks,
Jay
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
You could have just turned auth-calls to false and context to default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension>
Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext- rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@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
Brian West sip:brian@freeswitch.org
Aha. Thanks! :-) We're trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations. /b On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote: Thanks! :-) I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this? Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context. Thanks, Jay _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat You could have just turned auth-calls to false and context to default and accomplished the same thing ;) /b On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote: Sorry to bug you guys. I figured it out. In case anyone else is just learning to crawl with freeswitch. I enabled the following in the sip_profiles to get around the authorization errors (for now): <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/> <!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/> Then I started receiving a 404 route not found so I modified the public dialplan with the following: <extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension> Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider): <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. Ive been through the wiki and reviewed list archives but Im missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. Im sure its something obvious but what am I missing? Thanks, Jay _______________________________________________ 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 Brian West sip:brian@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 Brian West sip:brian@freeswitch.org
http://web.mac.com/brian.west/fs.jpg That is my ugly graph on how I setup the default config. /b On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote:
Aha… Thanks! J
We’re trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations.
/b
On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this?
Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context.
Thanks,
Jay
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
You could have just turned auth-calls to false and context to default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension>
Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext- rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@freeswitch.org
Thanks for all the assistance. BTW, when you clone default.xml to use as a base, do you assume the port 5060 or define a new port? Thanks! _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat http://web.mac.com/brian.west/fs.jpg That is my ugly graph on how I setup the default config. /b On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote: Aha Thanks! :-) Were trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations. /b On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote: Thanks! :-) I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this? Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context. Thanks, Jay _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat You could have just turned auth-calls to false and context to default and accomplished the same thing ;) /b On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote: Sorry to bug you guys. I figured it out. In case anyone else is just learning to crawl with freeswitch. I enabled the following in the sip_profiles to get around the authorization errors (for now): <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/> <!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/> Then I started receiving a 404 route not found so I modified the public dialplan with the following: <extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension> Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider): <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. Ive been through the wiki and reviewed list archives but Im missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. Im sure its something obvious but what am I missing? Thanks, Jay _______________________________________________ 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 Brian West sip:brian@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 Brian West sip:brian@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 Brian West sip:brian@freeswitch.org
define a new one On Apr 25, 2008, at 12:28 PM, Jay Reeder wrote:
Thanks for all the assistance. BTW, when you clone default.xml to use as a base, do you assume the port 5060 or define a new port?
Thanks!
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
http://web.mac.com/brian.west/fs.jpg
That is my ugly graph on how I setup the default config.
/b
On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote:
Aha Thanks! J
Were trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations.
/b
On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this?
Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context.
Thanks,
Jay
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
You could have just turned auth-calls to false and context to default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension>
Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext- rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@freeswitch.org
Should we post this on the wiki? :) -Dan From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 11:59 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat http://web.mac.com/brian.west/fs.jpg That is my ugly graph on how I setup the default config. /b On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote: Aha... Thanks! :) We're trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations. /b On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote: Thanks! :) I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this? Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context. Thanks, Jay ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat You could have just turned auth-calls to false and context to default and accomplished the same thing ;) /b On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote: Sorry to bug you guys. I figured it out. In case anyone else is just learning to crawl with freeswitch. I enabled the following in the sip_profiles to get around the authorization errors (for now): <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/> <!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/> Then I started receiving a 404 route not found so I modified the public dialplan with the following: <extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension> Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider): <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going. ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Problems with initial setup - basic nat Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal. Ive been through the wiki and reviewed list archives but Im missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. Im sure its something obvious but what am I missing? Thanks, Jay _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org<mailto: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 Brian West sip:brian@freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org<mailto: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 Brian West sip:brian@freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org<mailto: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 Brian West sip:brian@freeswitch.org
It should be there but I can't find it. (the pretty version) /b On Apr 25, 2008, at 4:35 PM, Daniel Hefti wrote:
Should we post this on the wiki? J
-Dan
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 11:59 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
http://web.mac.com/brian.west/fs.jpg
That is my ugly graph on how I setup the default config.
/b
On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote:
Aha… Thanks! J
We’re trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations.
/b
On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this?
Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context.
Thanks,
Jay
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
You could have just turned auth-calls to false and context to default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension>
Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext- rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@freeswitch.org
Do you mean this? http://wiki.freeswitch.org/wiki/Image:FreeswitchDiagram.png Ivan Den 25. april. 2008 kl. 23:47 skrev Brian West:
It should be there but I can't find it. (the pretty version)
/b
On Apr 25, 2008, at 4:35 PM, Daniel Hefti wrote:
Should we post this on the wiki? J
-Dan
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 11:59 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
http://web.mac.com/brian.west/fs.jpg
That is my ugly graph on how I setup the default config.
/b
On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote:
Aha… Thanks! J
We’re trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations.
/b
On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this?
Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context.
Thanks,
Jay
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Friday, April 25, 2008 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with initial setup - basic nat
You could have just turned auth-calls to false and context to default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/ $1"/> </condition> </extension>
Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext- rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Jay Reeder Sent: Thursday, April 24, 2008 4:40 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@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
The blind-auth is the important one here. sipX always sends an auth header even before we challenge it so by nature when FeeeSWITCH sees an auth header, it automaticly tries to qualify it in the db. I added that blind-auth option just for someone who had the same problem. When set at the same time as auth-cals is false it says to ignore the auth header. Can we maybe take this thread and make it into a wiki page on http://wiki.freeswitch.org On Sat, Apr 26, 2008 at 1:09 AM, Ivan C Myrvold <ivan@myrvold.org> wrote:
Do you mean this? http://wiki.freeswitch.org/wiki/Image:FreeswitchDiagram.png
Ivan
Den 25. april. 2008 kl. 23:47 skrev Brian West:
It should be there but I can't find it. (the pretty version) /b
On Apr 25, 2008, at 4:35 PM, Daniel Hefti wrote:
Should we post this on the wiki? J
-Dan
*From:* freeswitch-users-bounces@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org<freeswitch-users-bounces@lists.freeswitch.org> ] *On Behalf Of *Brian West *Sent:* Friday, April 25, 2008 11:59 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Problems with initial setup - basic nat
http://web.mac.com/brian.west/fs.jpg
That is my ugly graph on how I setup the default config.
/b
On Apr 25, 2008, at 11:40 AM, Jay Reeder wrote:
Aha… Thanks! J
We're trying to do outbound calling from behind nat. So the proper configuration is to still call through the default.xml (port 5060) and it would call OUT on nat.xml (port 5070)? In that case, what is outbound.xml (port 5080) used for? Would it be for MWI and strictly freeswitch->out applications? ------------------------------ *From:* freeswitch-users-bounces@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org<freeswitch-users-bounces@lists.freeswitch.org> ] *On Behalf Of *Brian West *Sent:* Friday, April 25, 2008 12:25 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Problems with initial setup - basic nat
Well first off you wouldn't use nat.xml for that.. you would clone default.xml and use it as a base. nat.xml is for OUTBOUND calling from behind nat only in the default config. its not designed to have inbound calls to it nor is it for registrations.
/b
On Apr 25, 2008, at 11:22 AM, Jay Reeder wrote:
Thanks! J
I did have auth-calls set to false in nat.xml but it wasnt working. Is there some other place I should have set this?
Whats the difference/application/use of the sample public context versus the default one? The sample nat.xml uses the public context.
Thanks,
Jay
------------------------------ *From:* freeswitch-users-bounces@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org<freeswitch-users-bounces@lists.freeswitch.org> ] *On Behalf Of *Brian West *Sent:* Friday, April 25, 2008 12:01 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Problems with initial setup - basic nat
You could have just turned auth-calls to false and context to default and accomplished the same thing ;)
/b
On Apr 25, 2008, at 10:55 AM, Jay Reeder wrote:
Sorry to bug you guys. I figured it out.
In case anyone else is just learning to crawl with freeswitch.
I enabled the following in the sip_profiles to get around the authorization errors (for now):
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) --> <param name="accept-blind-auth" value="true"/>
Then I started receiving a 404 route not found so I modified the public dialplan with the following:
<extension name="public_call"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/gateway/gafachi/$1"/> </condition> </extension>
Then I wasnt getting 2-way audio so I changed the sip profile for nat (which Im using internally) and set the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the sip-ip (since Im only using for internal nat through firewall to sip provider):
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/>
And now I have calls routed by sipx to freeswitch and through the firewall to our internet sip provider. Obviously the current configuration isnt secure but its enough to get things going.
------------------------------ *From:* freeswitch-users-bounces@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org<freeswitch-users-bounces@lists.freeswitch.org> ] *On Behalf Of *Jay Reeder *Sent:* Thursday, April 24, 2008 4:40 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Problems with initial setup - basic nat
Were setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. Ive wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
Ive been through the wiki and reviewed list archives but Im missing something.
I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples.
This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewalls external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch.
This is where Im stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch wont accept the call and is logging:
2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a
The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled.
Im sure its something obvious but what am I missing?
Thanks,
Jay _______________________________________________ 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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@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
Brian West sip:brian@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
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (5)
-
Anthony Minessale
-
Brian West
-
Daniel Hefti
-
Ivan C Myrvold
-
Jay Reeder