Yes thanks Lonnie

No the call never gets to the IP Phone. I manage all my forwarding within the 
Asterisk dial plan. And yes Im always keeping Asterisk in the path but as 
prompted by David, I suspect now that Asterisk is not bridging the call as I 
never actually Answer it in my dial plan.

We will see.

Regards
Michael Knill

-----Original Message-----
From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
Date: Tuesday, 14 March 2017 at 12:51 am
To: AstLinux List <astlinux-users@lists.sourceforge.net>
Subject: Re: [Astlinux-users] Astlinux on the edge

Michael,

I hope others here will offer their SIP experiences, but can you define in more 
detail what the failure mode is.  I'll guess a little ...

A call comes in via your SIP trunk provider, dials a local extension, either 
the extensions is busy (or DND set) or no answer then the Asterisk dialplan 
does what ?

Or are you using a "feature" of the IP Phone to initiate the outbound call when 
DND or other is set ?  Using Asterisk as the server or directly to the SIP 
trunk provider ?

Explain exactly who does what and when.

Bottom line, when behind NAT keep Asterisk in the path at all times.  Possibly 
in your failure case your IP Phone is re-inviting around Asterisk ?

Lonnie


On Mar 13, 2017, at 4:32 AM, Michael Knill <michael.kn...@ipcsolutions.com.au> 
wrote:

> Ok my initial NAT testing is exhibiting the following issue which I remember 
> previously occurred.
> Calls to and from extensions to external are fine with the below 
> configuration.
> The failure scenario however is an incoming call forwarding out to an 
> external call (hair pin) where there is no audio both ways.
> 
> I spend ages trying to troubleshoot the issue to no avail. I looked though 
> all the SIP SDP trying to work out what is happening. What I don't quite 
> understand, and I am hoping all the SIP experts can help, is that I don't 
> have any ALG’s set up so how does the external proxy know what media port to 
> connect to? I understand that rport is sent in the Via header which gives the 
> external address but this seems like its only for signalling!
> 
> What is interesting is that I do a packet sniff on the router external 
> interface (Mikrotik) and I don't see ANY RTP packets hitting or exiting. What 
> is also interesting is that when I answer the incoming call from an extension 
> and transfer it externally, the media works fine. 
> I suspect it has something to do with this which I cant seem to find any info 
> on:
> -- Local/0400113919@DialPlan1-00000025;2 requested media update control 20, 
> passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- SIP/gwy2-00000037 is making progress passing it to 
> Local/0400113919@DialPlan1-00000025;2
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
>    -- Local/0400113919@DialPlan1-00000025;2 requested media update control 
> 20, passing it to SIP/gwy2-00000037
> 
> Any ideas? No NAT for me currently until I can fix this.
> 
> Regards
> Michael Knill
> 
> -----Original Message-----
> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
> Date: Thursday, 9 March 2017 at 1:22 am
> To: AstLinux List <astlinux-users@lists.sourceforge.net>
> Subject: Re: [Astlinux-users] Astlinux on the edge
> 
> Michael,
> 
> If you place AstLinux behind a NAT firewall as a PBX ...
> 
> -- No NAT port forwarding to AstLinux (except for possible OpenVPN for remote 
> IP Phones) and disable any upstream SIP ALG's.
> 
> -- Set "directmedia=no" for all phones and the trunk, all media goes through 
> Asterisk
> 
> -- Set "qualify=yes" on trunk SIP peer to keep the upstream firewall state 
> active
> 
> -- Set "nat=force_rport,comedia" on the trunk SIP peer to force NAT handling, 
> the only peer that does NAT to Asterisk
> 
> -- Set "localnet=192.168.0.0/255.255.0.0' and "localnet=10.0.0.0/255.0.0.0" 
> to cover any LAN and OpenVPN networks which are not NAT'ed to Asterisk.
> 
> -- When using remote IP Phones over OpenVPN, since asterisk will bind to the 
> openvpn server tun interface, use the openvpn network (possibly 10.8.0.0/24) 
> for tunneled SIP endpoints.
> 
> (Readers, if I have missed or mangled any of the above, please correct.)
> 
> Bottom line, an AstLinux PBX behind NAT should be workable for production.
> 
> Lonnie
> 
> 
> On Mar 7, 2017, at 8:01 PM, Michael Knill <michael.kn...@ipcsolutions.com.au> 
> wrote:
> 
>> Hi thanks Lonnie. Sorry this went into my junk for some reason.
>> 
>> 1) Yes this is certainly a problem but I have also experienced problems with 
>> no media on calls being hairpinned through Asterisk from the external trunk. 
>> This may be solvable with port forwarding however. Maybe I should do some 
>> testing on this and specify some known and working router/firewall 
>> configurations.
>> 2) I use Open VPN for my external phones so it could be solved this way.
>> 
>> I am currently negotiating with the partner and it looks like they will take 
>> option 3 below which I think is the best compromise.
>> 
>> Regards
>> Michael Knill
>> 
>> -----Original Message-----
>> From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>
>> Reply-To: AstLinux List <astlinux-users@lists.sourceforge.net>
>> Date: Saturday, 4 March 2017 at 2:54 pm
>> To: AstLinux List <astlinux-users@lists.sourceforge.net>
>> Subject: Re: [Astlinux-users] Astlinux on the edge
>> 
>> Hi Michael,
>> 
>> My guess is "it depends" ... your IT partners go into a auto repair shop 
>> with a 5 year old residential-grade router, etc. (ie. a mess) then making 
>> AstLinux the edge device would be a major upgrade, not to mention the added 
>> voice functionality.
>> 
>> Then again your IT partners go into a dentist's office which were previously 
>> sold more router than they needed, it may not seem right to put AstLinux in 
>> front of it.
>> 
>> My guess is you need to plan for both situations.
>> 
>> A couple comments ...
>> 
>> 1) If AstLinux will only serve SIP endpoints on the private side, no roaming 
>> public endpoints, then being behind NAT is workable, only the trunk is 
>> effected by NAT.  Always disable any upstream SIP ALG's, almost always bad 
>> news.  Keep in mind no upstream port-forwarding is needed for this scenario, 
>> and always keep the AstLinux firewall enabled for the Adaptive Ban and other 
>> protections to be kept in place.
>> 
>> 2) Else if roaming public endpoints need to be supported, placing AstLinux 
>> at the edge will make SIP easier. AstLinux comes with a dmz-dnat plugin, the 
>> idea is to move a pre-existing router from the WAN to AstLinux's LAN with a 
>> static IP address and configure the plugin which internally performs a  " -j 
>> DNAT --to-destination $DMZ_IP " *all* traffic not allowed directly into 
>> AstLinux.  WARNING - this plugin was written many years ago and has not been 
>> tested as thoroughly as I would like to see for production purposes.  Though 
>> if there are issues with the dmz-dnat plugin they could be remedied.
>> 
>> Lonnie
>> 
>> 
>> On Mar 3, 2017, at 4:50 PM, Michael Knill 
>> <michael.kn...@ipcsolutions.com.au> wrote:
>> 
>>> Hi all
>>> 
>>> Im looking to push my Astlinux business this year and this will rely 
>>> heavily on partners. These partners will usually be IT Service providers 
>>> that have a number of small business customers and that they want to add 
>>> voice as a value add product.
>>> 
>>> Now here is where the problem lies. Most of these providers would currently 
>>> be maintaining the site firewall but as Astlinux is designed to be on the 
>>> edge, its an issue. So what do you do?
>>> 1)       Put Astlinux in front of their firewall and open up the necessary 
>>> ports and protocols. The problem here is that they lose flexibility in what 
>>> they can do and there is another provider in the mix. Its also a problem if 
>>> they are retailing the broadband connection for the site with too many 
>>> dependencies.
>>> 2)       Put their firewall on an Astlinux DMZ with a public IP Address. 
>>> They now have more flexibility and I can control Qos. Still issues with 
>>> being reliant on another provider and additional IP Addresses can be 
>>> expensive or unobtainable. I assume I can actually do this with Astlinux!
>>> 3)       Put Astlinux as a DMZ in their firewall with a public IP Address. 
>>> They now have complete control however QoS would need to be configured on 
>>> the firewall and additional IP Addresses can be expensive or unobtainable. 
>>> PS this is the model I have with one of my partners
>>> 4)       Sit behind the firewall and rely on port forwarding and/or ALG’s. 
>>> Inviting trouble but possible if you have a known working configuration
>>> 
>>> Im interested to know what others are doing in this space.
>>> 
>>> Regards
>>> Michael Knill
>> 
>> 
>> 
>> 
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> 
> 
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> 
> ------------------------------------------------------------------------------
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