Debug console output: https://pastebin.freeswitch.org/24572

My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.

>Is your endpoint registered? How often is it set to reregister?

Yes, and the zoiper default registration expiry is 3600

 

>Does your router have a sip alg? Are the ports opened in your firewall?

> ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter).  No firewall ports are explicitly open, but allow bi-directional with existing states.  I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts.  I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed.  Also, unfortunately, I cannot get a static ip.

Thanks!

  -Brian

On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max@nysolutions.com> wrote:

It is probably sending call to your extension@domainname, if your external ip is your domain name then you will see the external ip.

Is your endpoint registered? How often is it set to reregister?

Does your router have a sip alg? Are the ports opened in your firewall?

Very hard to guess without a log of a call with debug enabled.

sofia global sip trace on

 

Thanks,

 

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US
Service Request Email: support@nysolutions.com

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

 

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

 

Hello all!

  I'm new to freeswitch, so I'm sure this is just a newbie configuration error.  Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages.  I've already followed the confluence page on configuring NAT.

 

My Setup:

Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.

SIP Provider: Flowroute

 

NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute.  I configured external sip and rtp to use stun entries.  External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.  

 

I have one DID configured to go directly to my one extension.  My extension can register just fine.  My extension can dial out.  When I call my cell, the call connects and I have 2 way audio.  When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).

 

When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?

 

I see this line in the console:  2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072

 

 

Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension>  ?

 

Sofia Status says:

 

                     Name        Type                                      Data  State

=================================================================================================

            external-ipv6   profile                   sip:mod_sofia@[::1]:5080            RUNNING (0)

                 external      profile           sip:mod_sofia@<external_ip>:5080     RUNNING (0)

external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com          REGED

            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)

                 internal       profile            sip:mod_sofia@<internal_ip>:5060    RUNNING (0)

=================================================================================================

 

If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue?  I feel like it's just me missing something, I just can't determine what that might be.

 

Thanks,

  -Brian

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