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[steve.d.ward at gmail.com]

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 dialplan I'm trying to do a bridge w/ something like "sofia/gateway/${default_gateway}/12345." 
 
When I try to make the call I see from the console:
 
... New Channel sofia/external/12345 ...
... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!]
... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER]
 
The Originate fails.
 
I tried sticking to what the instructions laid out for this in the Connecting FS and Asterisk wiki page, so I'd appreciate some help in figuring out what's going on.  Thanks.
 
 
 

[msc at freeswitch.org]

2009/3/16 Steven Ward <steve.d.ward@gmail.com>:

[steve.d.ward at gmail.com]

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, 2009 at 11:59 AM, Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> wrote:

[msc at freeswitch.org]

2009/3/16 Steven Ward <steve.d.ward@gmail.com>:

[mrene_lists at avgs.ca]

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:

[brian at freeswitch.org]

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:

[steve.d.ward at gmail.com]

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/1000@pbx-sip-3.usa.internal.net ([email]sofia/internal/1000@pbx-sip-3.usa.internal.net[/email]) Execute bridge(sofia/gateway/${default_gateway}/70904)
2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 switch_core_session_exec() sofia/internal/1000@pbx-sip-3.usa.internal.net ([email]sofia/internal/1000@pbx-sip-3.usa.internal.net[/email]) Expanded String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/70904)
2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid Gateway
2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close Channel N/A [CS_NEW]
2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 switch_ivr_originate() Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed.  Cause: INVALID_NUMBER_FORMAT


What else am I missing and not doing right?  Thanks again for your help.

[steve.d.ward at gmail.com]

Sure thing.  Here it is:
 
<include>
  <user id="$${default_provider}">
      <gateway name="$${default_provider}">
        <param name="username" value="$${default_provider_username}"/>
        <param name="password" value="$${default_provider_password}"/>
        <param name="from-user" value="$${default_provider_username}"/>
        <param name="from-domain" value="$${default_provider_from_domain}"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="$${default_provider_register}"/>
        <param name="retry-seconds" value="30"/>
        <param name="extension" value="$${default_provider_contact}"/>
        <param name="contact-params" value="domain_name=$${domain}"/>
        <param name="context" value="default"/>
      </gateway>
  </user>
</include>

In vars.conf I supplied the variables' values:
  <X-PRE-PROCESS cmd="set" data="default_provider=pbx-sip-4.usa.internal.net"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=pbx-sip-4"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=9997"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=pbx-sip-4.usa.internal.net"/>
  <!-- true or false -->
  <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_contact=1000"/>


 
2009/3/16 Brian West <brian@freeswitch.org (brian@freeswitch.org)>

[brian at freeswitch.org]

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:

[msc at freeswitch.org]

2009/3/16 Steven Ward <steve.d.ward@gmail.com>:

[steve.d.ward at gmail.com]

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 Brian West <brian@freeswitch.org (brian@freeswitch.org)>

[steve.d.ward at gmail.com]

Thanks.
 
I created the gateway file in conf/directory/default/


 
On Mon, Mar 16, 2009 at 1:24 PM, Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> wrote: