[Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Thu Feb 25 19:12:36 MSK 2016

Does internal call also go direct to voicemail?
Are you sure you don’t have a *99 before the extension number?

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
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From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Chow
Sent: Thursday, February 25, 2016 11:00 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?

Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension?
On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow at digitaldescent.net<mailto:brianchow at digitaldescent.net>> wrote:

Ok, in that case, what do I look at next to see why it goes straight to voicemail instead of ringing my extension?
On Feb 25, 2016 5:52 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Its just the channel name so don't get tripped up by that, its just named with some default data about the session.

On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch at digitaldescent.net<mailto:freeswitch at digitaldescent.net>> wrote:
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 at nysolutions.com<mailto:max at nysolutions.com>> wrote:
It is probably sending call to your extension at 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<tel:212.400.7650> 888.IPPBX.US<http://888.IPPBX.US>
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.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 at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian Chow
Sent: Wednesday, February 24, 2016 10:00 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at 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<sip:mod_sofia@%3cexternal_ip%3e:5080>     RUNNING (0)
external::<uuid>          gateway                      sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com>          REGED
            internal-ipv6    profile                   sip:mod_sofia@[::1]:5060            RUNNING (0)
                 internal       profile            sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e: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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Brian West
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_________________________________________________________________________
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