On Sat, 2016-05-28 at 18:59 +1000, Andrew Pam wrote:

...

I came across a few forum posts and discussions about the Internode settings, but I wonder if anyone from the LUV community runs such a setup or has some experience with Asterisk and Internode (especially their Nodephone product)?

I've been using Nodephone for many years ... [...] I've thought about setting up an Asterix or FreePBX server but have never got around to it because frankly I mostly use my Nodemobile instead.

Hi Andrew, hi Jason (and everyone else interested), Thanks for your replies guys. Meanwhile I installed Asterisk and FreePBX on a Debian 8 machine and configured NodePhone as a trink without any issues. In fact it was pretty straight forward and I look forward demonstrating the current state to a small team in our office later this week. In my tests I have used several clients (Linux and Windows based as well as an Android smartphone). Let's see how the default setup works in real-life scenarios :-) Cheers Michael

On Mon, 2016-06-13 at 21:05 +1000, Russell Coker wrote:

On Mon, 13 Jun 2016 08:47:34 PM Michael Schams via luv-main wrote:

...

In fact it was pretty straight forward and I look forward demonstrating the current state to a small team in our office later this week. In my

This might be a good topic for a talk at a future LUV meeting. We haven't had a talk about VOIP for a while.

If you are aiming to me: I don't pretend to be an Asterisk/FreePBX/VoIP expert by just reading some HowTos and following tutorials on the Internet :-) Everyone can do this and as I pointed out before, it's really easy and pretty straight forward. This could be a talk at the LUV Beginners meetup or maybe a workshop? If you are aiming to the broader community: Personally, I would appreciate if someone with more knowledge and expertise would give a talk about Asterisk/FreePBX/VoIP. Maybe someone who has set up this kind of systems more than once and has experience? I would definitely be in the audience ;-) Cheers Michael

On Mon, Jun 13, 2016 at 10:29 PM, Andrew McGlashan wrote:

Setting up Asterix or FreePBX or anything similar is not something that should be done lightly. VoIP providers lose an awful lot of money if there are any loop holes in their setup; perhaps even just a weak password. So, it is a serious risk situation, potentially; especially when there are continual software updates to fix vulnerabilities in all kinds of software.

I'm not saying don't do it, but I am saying that you have to understand the risks and perhaps you would be better off not doing it.

Hi Andrew, can you elaborate a bit about Asterisk/FreePBX security issues? My general observation about proprietary PABXes I found so far: - they get serviced by contractors - passwords get never changed (mainly to make access "easier", and/or because the different built-in accounts and roles are not understood and so all management is done with the most powerful account) - patching only happens when a feature cannot be used with the old version (e.g. a reporting or configuration tool requires a more recent version). In fact I have never experienced a continuously serviced and upgraded PABX in my work experience. So, as long as you follow the same bad practice, the main difference seems to be that your manager blames you instead of the servicing company;-) Advantages of using Asterisk or similar: - You can have two of them to test configuration changes,patching etc. - You can snapshot them - You get regular updates as all other systems in a setup that is using the same OS/distribution - You may have a better understanding about the security risk because you are aware of it, know a bit of least privileged access etc. IMHO the risk is more on the management side: managers without technical understanding do not care about what behind the scenes so capable staff is not able to establish best practice and critical maintenance does not happen. Regards Peter

Hi, On 14/06/2016 10:29 AM, Peter Ross via luv-main wrote:

On Mon, Jun 13, 2016 at 10:29 PM, Andrew McGlashan wrote:

Setting up Asterix or FreePBX or anything similar is not something that should be done lightly. VoIP providers lose an awful lot of money if there are any loop holes in their setup; perhaps even just a weak password. So, it is a serious risk situation, potentially; especially when there are continual software updates to fix vulnerabilities in all kinds of software.

I'm not saying don't do it, but I am saying that you have to understand the risks and perhaps you would be better off not doing it. Hi Andrew,

can you elaborate a bit about Asterisk/FreePBX security issues?

I think if you are very careful with passwords and access generally, then you should be mostly fine. The trouble comes when an account gets compromised and it has services behind it that will cost you money; they then start calling expensive services that can cost you a fortune. If you have a closed system, then there is less risk too. ISPs and VoIP providers go to extremes to limit risk after they've been bitten; not allowing services to work from different geographic locations is part of their mitigation. I don't know of any specific current issues or any recent compromises. But, I do say, be careful, it could be very expensive if you get something wrong; an open relay with a mail server would be a problem, but an open VoIP service can be real expensive trouble (potentially). Cheers A.

On 16/06/2016 11:35 PM, Michael Schams via luv-main wrote:

But I get your point Andrew :-) I am not working for a VoIP provider and following some HowTos and tutorials on the Internet does not make me an Asterisk/FreePBX/VoIP expert. I assume, I can trust our internal staff.

Unfortunately, sometimes internal staff can be the problem when nothing else is. Cheers A.

Jason White via luv-main <luv-main@luv.asn.au> writes:

I've used Nodephone successfully with FreeSWITCH using a Snom 320 and an Android phone as SIP clients.

This configuration is no longer current, however, as circumstances have changed. Nevertheless, the Nodephone account is still serving the needs of family members in Melbourne.

Just wondered if anybody here had any ideas of a problem I have had with Freeswitch. I have asked about this on the mailing list, and reported a bug, but nobody seems to be able to help me. If I get an external phone call, and it redirects to voicemail, the voicemail app does not seem to receive any audio. It then responds immediately by saying the message is too short. This works fine if I make the phone call from an internal SIP phone. Also works fine: audio from the external phone if I use the echo test, or if the phone call is answered normally on a SIP extension. So I think this rules out firewall issues, NAT issues (note: there is no NAT), etc. The Codec for the transmitted/recived data is PCMA, looks good to me. I think some people looking at the logs in my bug report may have got confused with some of the messages. Thought maybe it was a problem with my SIP provider (not that that makes any sense), but reproduced the exact same problem with NodePhone. Out of desperation, I copied almost the entire external sip profile to the internal sip profile settings, but internal still works fine. Any ideas? Thinking I might need to ditch FreeSwitch for something that actually works :-( -- Brian May <brian@linuxpenguins.xyz> https://linuxpenguins.xyz/brian/

Brian May <brian@linuxpenguins.xyz> wrote:

Just wondered if anybody here had any ideas of a problem I have had with Freeswitch.

Try their IRC channel; you may be able to contact the developers there. In my experience, they resolve problems quickly.