[Freeswitch-users] Phones registered to internal profile hit external profile when calling

Donny Hardyanto hardyanto.donny at gmail.com
Wed Oct 23 04:55:49 MSD 2013


<param name="context" value="public"/>

I think that the culprit if this internal.xml

Donny
On Oct 22, 2013 8:36 PM, "hcoin" <hcoin at quietfountain.com> wrote:

>  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.
>
> No <aliases>, and       <domains>  <domain name="all" alias="true"
> parse="false"/> </domains>
>
>
>         <param name="context" value="public"/>
>         <param name="rfc2833-pt" value="101"/>
>         <param name="sip-port" value="5090"/>
>         <param name="dialplan" value="XML"/>
>         <param name="dtmf-duration" value="2000"/>
>         <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>         <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>         <param name="rtp-timer-name" value="soft"/>
>         <param name="rtp-ip" value="$${local_ip_v4}"/>
>         <param name="sip-ip" value="$${local_ip_v4}"/>
>         <param name="stun-enabled" value="false"/>
>         <param name="hold-music" value="$${hold_music}"/>
>         <param name="apply-inbound-acl" value="domains"/>
>         <param name="local-network-acl" value="rfc1918.auto"/>
>         <param name="auth-calls" value="false"/>
>         <param name="inbound-reg-force-matching-username" value="true"/>
>         <param name="auth-all-packets" value="false"/>
>         <param name="record-path" value="$${recordings_dir}"/>
>         <param name="record-template"
> value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>         <param name="manage-presence" value="true"/>
>         <param name="presence-hosts" value="pbx.mamabosso.com,
> pbx.amfoodstyles.com"/>
>         <param name="presence-privacy" value="$${presence_privacy}"/>
>         <param name="inbound-codec-negotiation" value="generous"/>
>         <param name="tls" value="$${internal_ssl_enable}"/>
>         <param name="tls-only" value="false"/>
>         <param name="tls-bind-params" value="transport=tls"/>
>         <param name="tls-sip-port" value="5091"/>
>         <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
>         <param name="tls-passphrase" value=""/>
>         <param name="tls-verify-date" value="true"/>
>         <param name="tls-verify-policy" value="none"/>
>         <param name="tls-verify-depth" value="2"/>
>         <param name="tls-verify-in-subjects" value=""/>
>         <param name="tls-version" value="$${sip_tls_version}"/>
>         <param name="nonce-ttl" value="60"/>
>         <param name="rtp-timeout-sec" value="300"/>
>         <param name="rtp-hold-timeout-sec" value="1800"/>
>         <param name="challenge-realm" value="auto_from"/>
>         <param name="inbound-bypass-media" value="false"/>
>         <param name="rtp-autoflush" value="true"/>
>         <param name="rtp-autoflush-during-bridge" value="true"/>
>         <param name="suppress-cng" value="false"/>
>         <param name="rtp-rewrite-timestamps" value="false"/>
>         <param name="auto-jitterbuffer-msec" value="120"/>
>
> The external profile has alias=false in the domains, no aliases and
> parse=true
>
>
>
> On 10/22/2013 06:13 AM, Donny Hardyanto wrote:
>
> Is your client in the internet or the lan?
>
>  Donny
>
>
> On Tue, Oct 22, 2013 at 1:01 PM, hcoin <hcoin at quietfountain.com> wrote:
>
>>
>> This has been a really frustrating problem, I'm sure the answer is
>> simple but I just can't see it.
>>
>> I had several extensions registered to the internal profile, sending
>> calls out the external profile to a sip-pstn gateway, all seemed fine.
>>
>> Then created another internal profile, using a different sip port on the
>> same lan address, because of 'no device left behind' and NAT issues..
>>
>> All seemed well, all the phones register normally.   Looking at the
>> databases in FS they all show the proper ports, the proper domains, etc.
>>
>> However, every single call gets picked up as a new call via
>> sophia/external/... and it hits the public dialplan normally -- except
>> that's the wrong plan, it should hit the default plan and be identified
>> as sofia/internal/.... and so forth.
>> 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel
>> sofia/external/hcoin at pbx.foobar.com[28ed125a-3adb-11e3-9cc1-cbb8efb09b83]
>>
>> What could possibly be the reason phones registered on the internal
>> profile have their new calls identified as sophia/external and don't hit
>> the correct plan?  Both the phones and the freeswitch are on the same
>> subnet.  This should be so vanilla.  What am I missing?
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www..freeswitchsolutions.com <http://www.freeswitchsolutions.com>
>>
>> 
>> 
>>
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>
>
>
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>
> 
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>
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