Michael,

If you have a spare AstLinux box laying around, replace the Mikrotik with 
AstLinux just for the edge NAT firewall as a test, and to make sure asterisk is 
not interfering:
-- user.conf --
ASTERISK_DAHDI_DISABLE="yes"
--
on the spare edge AstLinux box.

If that works the finger is pointed to the Mikrotik.

Lonnie


On Mar 14, 2017, at 6:05 PM, Michael Knill <michael.kn...@ipcsolutions.com.au> 
wrote:

> Well the problem still exists. I actually do answer the call and even if I 
> Answer it right at the beginning its still has issues with ‘requested media 
> update control 20’ and no media.
> 
> Yes it may have something to do with the SIP Provider but I simply cannot get 
> this to work.
> I have added the following to sip.conf:
> nat=force_rport,comedia
> externaddr=121.45.222.129
> localnet=192.168.0.0/255.255.0.0 
> localnet=10.0.0.0/255.0.0.0
> localnet=172.16.0.0/12
> localnet=169.254.0.0/255.255.0.0
> 
> I tried with and without the externaddr!
> 
> Any other ideas?
> 
> 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 4:01 am
> To: AstLinux List <astlinux-users@lists.sourceforge.net>
> Subject: Re: [Astlinux-users] Astlinux on the edge
> 
> Michael,
> 
> Keeping Asterisk in the path is key, and calling Answer() is required at some 
> point to do that.
> 
> I always call Answer() before calling local phones, of course any IVR 
> requires calling Answer() first.
> 
> Though it may be possible, depending on your SIP trunk provider and enabling 
> "directmedia=yes" for the trunk only, to selectively re-invice inbound calls 
> back to the SIP trunk and not calling Answer().  Since this depends on your 
> SIP trunk provider, it may work one day and stop working another day.
> 
> If these kind of "hair-pin" calls are not common, play it safe and answer the 
> call and dial back out, keeping Asterisk in the path.
> 
> Lonnie
> 
> 
> On Mar 13, 2017, at 11:33 AM, Michael Knill 
> <michael.kn...@ipcsolutions.com.au> wrote:
> 
>> 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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>> projects. Get started today and enter our developer competition.
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>> pay...@krisk.org.
>> 
>> 
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>> pay...@krisk.org.
> 
> 
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> 
> 
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