<p dir="ltr"><param name="context" value="public"/></p>
<p dir="ltr">I think that the culprit if this internal.xml</p>
<p dir="ltr">Donny</p>
<div class="gmail_quote">On Oct 22, 2013 8:36 PM, "hcoin" <<a href="mailto:hcoin@quietfountain.com">hcoin@quietfountain.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>Lan. Lan registered phones can't even
get a tone stream. Phones register promptly, everything seems in
order. It's just that every call coming in on the lan interface
and 'internal' port gets identified as 'sofia/external/ and so
never hits the default dial plan. Here's a bit of the internal
profile. The site has some vpns so anything in the rfc1918 space
has no nat. Shouldn't matter anyway as all extensions on the same
lan subnet as the fs box get identified as 'external' as well.<br>
<br>
No <aliases>, and <domains> <domain
name="all" alias="true" parse="false"/> </domains><br>
<br>
<br>
<param name="context" value="public"/><br>
<param name="rfc2833-pt" value="101"/><br>
<param name="sip-port" value="5090"/><br>
<param name="dialplan" value="XML"/><br>
<param name="dtmf-duration" value="2000"/><br>
<param name="inbound-codec-prefs"
value="$${global_codec_prefs}"/><br>
<param name="outbound-codec-prefs"
value="$${global_codec_prefs}"/><br>
<param name="rtp-timer-name" value="soft"/><br>
<param name="rtp-ip" value="$${local_ip_v4}"/><br>
<param name="sip-ip" value="$${local_ip_v4}"/><br>
<param name="stun-enabled" value="false"/><br>
<param name="hold-music" value="$${hold_music}"/><br>
<param name="apply-inbound-acl" value="domains"/><br>
<param name="local-network-acl"
value="rfc1918.auto"/><br>
<param name="auth-calls" value="false"/><br>
<param name="inbound-reg-force-matching-username"
value="true"/><br>
<param name="auth-all-packets" value="false"/><br>
<param name="record-path"
value="$${recordings_dir}"/><br>
<param name="record-template"
value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/><br>
<param name="manage-presence" value="true"/><br>
<param name="presence-hosts"
value="<a href="http://pbx.mamabosso.com" target="_blank">pbx.mamabosso.com</a>,<a href="http://pbx.amfoodstyles.com" target="_blank">pbx.amfoodstyles.com</a>"/><br>
<param name="presence-privacy"
value="$${presence_privacy}"/><br>
<param name="inbound-codec-negotiation"
value="generous"/><br>
<param name="tls" value="$${internal_ssl_enable}"/><br>
<param name="tls-only" value="false"/><br>
<param name="tls-bind-params"
value="transport=tls"/><br>
<param name="tls-sip-port" value="5091"/><br>
<param name="tls-cert-dir"
value="$${internal_ssl_dir}"/><br>
<param name="tls-passphrase" value=""/><br>
<param name="tls-verify-date" value="true"/><br>
<param name="tls-verify-policy" value="none"/><br>
<param name="tls-verify-depth" value="2"/><br>
<param name="tls-verify-in-subjects" value=""/><br>
<param name="tls-version"
value="$${sip_tls_version}"/><br>
<param name="nonce-ttl" value="60"/><br>
<param name="rtp-timeout-sec" value="300"/><br>
<param name="rtp-hold-timeout-sec" value="1800"/><br>
<param name="challenge-realm" value="auto_from"/><br>
<param name="inbound-bypass-media" value="false"/><br>
<param name="rtp-autoflush" value="true"/><br>
<param name="rtp-autoflush-during-bridge"
value="true"/><br>
<param name="suppress-cng" value="false"/><br>
<param name="rtp-rewrite-timestamps" value="false"/><br>
<param name="auto-jitterbuffer-msec" value="120"/><br>
<br>
The external profile has alias=false in the domains, no aliases
and parse=true<br>
<br>
<br>
<br>
On 10/22/2013 06:13 AM, Donny Hardyanto wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Is your client in the internet or the lan?
<div><br>
</div>
<div>Donny</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Tue, Oct 22, 2013 at 1:01 PM, hcoin
<span dir="ltr"><<a href="mailto:hcoin@quietfountain.com" target="_blank">hcoin@quietfountain.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
This has been a really frustrating problem, I'm sure the
answer is<br>
simple but I just can't see it.<br>
<br>
I had several extensions registered to the internal profile,
sending<br>
calls out the external profile to a sip-pstn gateway, all
seemed fine.<br>
<br>
Then created another internal profile, using a different sip
port on the<br>
same lan address, because of 'no device left behind' and NAT
issues..<br>
<br>
All seemed well, all the phones register normally. Looking
at the<br>
databases in FS they all show the proper ports, the proper
domains, etc.<br>
<br>
However, every single call gets picked up as a new call via<br>
sophia/external/... and it hits the public dialplan normally
-- except<br>
that's the wrong plan, it should hit the default plan and be
identified<br>
as sofia/internal/.... and so forth.<br>
2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034
New Channel<br>
sofia/external/<a href="mailto:hcoin@pbx.foobar.com" target="_blank">hcoin@pbx.foobar.com</a>
[28ed125a-3adb-11e3-9cc1-cbb8efb09b83]<br>
<br>
What could possibly be the reason phones registered on the
internal<br>
profile have their new calls identified as sophia/external
and don't hit<br>
the correct plan? Both the phones and the freeswitch are on
the same<br>
subnet. This should be so vanilla. What am I missing?<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
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<br>
<fieldset></fieldset>
<br>
<pre>_________________________________________________________________________
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