Re: [asterisk-users] New Asterisk build

[Glenn Geller (VDOPh)]

If you are really wanting to build something on Raspberry Pi or similar ARM
platform, you could also take a look at Elastix for ARM.

http://www.elastix.com/en/downloads/ Elastix is a fully integrated
platform, and includes the majority of necessary components in one
installation.

The new Raspberry Pi 2 platform may be perfect for your needs in this
respect, although based on your load, the B+ board may be more available at
this time, and slightly cheaper.

The Pi 2 is about double the core processing speed.

YMMV

Thanks,



*Glenn Geller*

*VDOTel*


On Fri, Mar 6, 2015 at 12:34 PM, John Novack SCII 
jnov...@stromberg-carlson.org wrote:

 Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux
 ( instructions at AstLinux.org ) Move your Digium card and your confs , fix
 up any differences from your older version of Asterisk to the fairly
 current version 11 currently available with AstLinux.
 Use the GUI to edit and mage the system, as AstLinux has a somewhat
 different directory structure than you may be familiar with
 You should be up and running in a couple of hours, have a low power  20
 watts, fanless flash based system that will just work in a real case.
 The Pi is OK for a playtoy and some testing, but I much prefer the HP thin
 clients for a robust installation.
 I assume you are not doing any fancy call center or heavy database work.
 For a home or home office it is a really good solution.
 AstLinux is also used with other embedded installations on computers with
 multiple Ethernet ports, acting as router and firewall in addition.
 I prefer the component solution personally, which makes the HP thin
 clients the way to go.


 John Novack


 I have built more than 30 of these systems on various HP Thin Clients,
 used off of eBay with no failures

 Ira wrote:

Hello Asterisk,
  Back in 2009 I built a small Intel Atom based computer running
Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs
line and six or so SIP numbers. So basically no load. I'm
feeling like it's time to build another machine. It's probably
silly, but it's been six years and I can't upgrade the OS
which is falling behind. I'd likely just put it on a Raspberry
Pi or something like that, but I need the one POTS line and
all I have for that at the moment is a Digium card for a PCI
slot.

Are there any current thoughts on this?
   -- Ira



 --
 Dog is my Co-Pilot



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Re: [asterisk-users] Trying to register Softpone in AWS Cloud

[Glenn Geller (VDOPh)]

Hello,

Unless you're directly connected to AWS via a physical cable, or other
direct connect method, you cannot use the internal IP AWS address from the
public Internet.

Either use the existing PUBLIC IP address (shown in the control panel for
the EC2 instance), or assign a new Elastic IP, and point this to the EC2
instance in question.

In order to work with NAT, you will need to modify your conf file with the
external IP address, so Asterisk can send this address in the headers for 2
way audio.

Good luck!

Thanks,

*Glenn Geller*


On Wed, Apr 15, 2015 at 5:27 AM, akhilesh chand omakhileshch...@gmail.com
wrote:

 Hi Folks,

 I'm trying to register softphone(3CX Phone) in AWS Cloud but I'm not able
 to register I got below screen.

 [image: Inline image 1]


 Register Screen for 3CX Phone


 [image: Inline image 1]



 Regards
 Akhilesh

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Re: [asterisk-users] FXO advice

[Glenn Geller (VDOPh)]

We also use the Obihai devices successfully in this type of configuration.

They also have a website that can function as your provisioning server if
you don;t have one yet.

They make inexpensive single port, and multi-port versions.

If you are new to this in general, I'd suggest your start with the Obi100
or 200 series.

Good luck!

Thanks,



*Glenn Geller*

*VDO-Ph International*
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Re: [asterisk-users] ST2030 replacement

[Glenn Geller (VDOPh)]

Also try vtech vsp725

Thanks,

*Glenn*


On Thu, Jan 7, 2016 at 8:35 AM, Sil  wrote:

> Hello,
> I am looking for a replacement for my Thomson ST2030SIP.
> My specifications are as follows :
> - 2 lines.
> - 6 BLF keys.
> - PoE.
> Can you give me a return on the models you use ?
> Thanks.
>
> Sil
>
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Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

[Glenn Geller (VDOPh)]

Hi,

Usually, the trunk provider(s) provide a mechanism to support this, and
it's the "Tech Prefix" or just "Prefix".

So, for Tenant 1, send 12348005551212 and for Tenant 2, send 56788005551212.

They'll then strip the prefix, and send along to 8005551212

Most trunk providers worth anything will support this type of termination.

Hope this helps,

*Glenn @ VDO*


On Wed, May 25, 2016 at 2:13 PM, Attila Megyeri 
wrote:

> Hi!
>
>
>
> I would like to reopen a discussion that I saw a couple of years ago, with
> the subject  “Sending Calls via SIP trunk from two different IP addresses
> from same Asterisk Machine”
>
>
>
> The use case is simpe: There are providers that want to see a separate
> source IP address for each of their customers SIP trunks. Now, if we have
> an asterisk box with several customers, we have a problem.
>
>
>
> Does anyone have experience in this topic? How could we send outgoing
> calls (to the same destination IP) from different source IPs depending on
> the caller ID (Based on From: field, sip account, preferred-identity,
> whatever).
>
>
>
> I was thinking about some Kamailio, or SBC that would take the calls from
> asterisk using a user/pass authentication on a single interface, and
> initiate calls from a dedicated IP address for each customer.
>
> Any better idea?
>
>
>
> Thanks
>
>
>
> Attila
>
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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Glenn Geller (VDOPh)]

As far as I know, Zoiper does support TLS across all platforms.

We've tested and deployed this on their "paid" software on Android and
Windows, and it works well.

Thanks,

*Glenn @ VDOTel*

On Wed, Feb 15, 2017 at 2:54 PM, Marcelo Terres  wrote:

> Zoiper?
>
> On 15 Feb 2017 6:46 p.m., "Motty Cruz"  wrote:
>
>> Hello, I have a user that prefers Soft SIP phone install on his laptop,
>> for security reasons I have enable TLS on our Asterisk server to support
>> TLS authentication, It works well with hard phones. Has anybody in this
>> forum use SIP Soft phones with TLS authentication enabled? Any suggestions?
>>
>>
>>
>> Thanks,
>> Motty
>>
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Re: [asterisk-users] Installing Asterisk on MAC native

[Glenn Geller (VDOPh)]

If you're looking for installing on a MAC, best to start searching for MAC
OSX install

See here:
http://www.voip-info.org/wiki/view/Asterisk+Getting+Started+on+MacOSX

I don't know how old this is, or if it directly applies to your task at
hand, but it may be a start.

Also, if you're just looking for a simple PBX for light usage, there may be
other options out there for MAC OS as well.

Good hunting!

*Glenn Geller*

On Tue, Sep 20, 2016 at 1:58 PM, Saint Michael  wrote:

> ​I need to install Asterisk on a MAC, native, no virtualization.
> Has anybody done this? Are there documents on the Internet?
> I googled it and all web sites that claimed to help installing Asterisk on
> a MAC have disappeared. Is it possible at all?
> Digium should actually have a MAC app in the Apple store with a PBX. It
> should be a paid app. I would buy it right away.
>
> ​
>
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Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK

[Glenn Geller (VDOPh)]

Hi Chris,

This would require either an external (or possible internal) GSM/SIM Voip
Gateway.

Google: "VOIP Gateway GSM Converter SIP IP Phone Adapter" or similar for
options. I see some for around $60 US

Basically, you'll want to send SIP invite(s), and other commands to and
from this device (probably on it's local address, for security, and it acts
just as any SIP provider would.

Thanks,


*Glenn @ VDO*


On Mon, Dec 5, 2016 at 6:36 PM, christopher kamutumwa <
chriskamutu...@gmail.com> wrote:

> Is it possible to have a simcard configured and become incoming line
> and outgoing on asterisk and also have the IVR function? If yes wat
> hardware is required to have this Accompished
>
> Thanks
>
>
> Chris
>
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Re: [asterisk-users] double NAT - one way audio

[Glenn Geller (VDOPh)]

Hi Andre,

On your comment "unfortunately there is no bridge mode or any comparable
mode available", sometimes the carrier (if it's a carrier supplied DSL
router) will have these settings hidden from standard user's eyes.

You may need to call your ISP and request them to place your DSL router
into "bridged" mode.

Thanks,


*Glenn @ VDOTel*


On Sun, Mar 12, 2017 at 11:57 PM, Andre Gronwald 
wrote:

> Hi Glenn,
> unfortunately there is no bridge mode or any comparable mode available. I
> am using the same router (but another type) on my private homenetwork with
> another router at the back (=> same architecture as in this failing
> scenario), but everything works fine.
> There are only two differences:
> 1. Another Type (w724v Type B instead of w724v Type A)
> 2. No VoIP services used by w724v (which is on Type A hardware currently
> the case, maybe disablling them helps?).
>
> I will check to switch 2., but that is not easily doable because there are
> productive numbers used... The asterisk installation is currently in
> development...
>
> regards,
> andre
>
> --
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> Andre Gronwald
> andregronwal...@gmail.com  >
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Re: [asterisk-users] double NAT - one way audio

[Glenn Geller (VDOPh)]

Hi Andre,

Some routers just simply won't support this double-nat scenario you
describe. Othera will... And without any special forwarding.

Is it possible to put the first router into "bridge" mode, and use the
second router as the actual NAT router?

This may be the quickest solution to your problems. Good luck!

Thanks, Glenn (mobile)


On Mar 11, 2017 8:50 AM, "Andre Gronwald"  wrote:

Hi all,

I have a setup which is not working right now:

Provider - DSL-Router (192.168.2.1) - Bintec-Router (10.17.46.66) -
Asterisk (10.17.46.99)

My issue: Everything works, but RTP is only going from my Asterisk towards
the provider. Asterisk is configured to use SIP-ports 55060 and RTP-ports
51000-51999.
Those ports are forwarded on DSL-router  to the bintec router and from the
bintec router to asterisk.

what I see is the Invite from provider goes to 192.168.2.1 and rtp port
7070. my asterisk responds with audio to be sent to ip address 80.142.12.12
port 51242.
Afterwards RTP goes from 10.17.46.99:51242 to 192.168.2.1:7070. but the RTP
back is not coming in.

I would expect the RTP traffic to be sent to 80.142.12.12 (intetnal
192.168.2.1) port 51242 - then i would have successfully two way audio.
But why is port 7070 used?

The DSL-router is a speedport w724v type A.

regards,
andre

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Re: [asterisk-users] Copying received and sent RTP packets due legal obligations

[Glenn Geller (VDOPh)]

I'd suggest you take a look at Voipmonitor, it may have what you need in
the community version. It's built for monitoring SIP/VoIP traffic.

Pretty inexpensive to license the reporting tool as well, if needed.

Also, you may need to have a switch/router that supports port mirroring.

Thanks,

*Glenn*

On Wed, Jul 12, 2017 at 2:43 PM, Mark Wiater 
wrote:

> On 7/12/2017 5:30 PM, Holger Freyther wrote:
> > I have to copy/mirror/forward the RTP streams for some selected call
> > to an external address/port
> I'd think that what you want to do might be best done outside of
> Asterisk.  If you're working with SIP, I'd suggest packet capture tools.
>
>
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Re: [asterisk-users] Streaming MoH from iHeart radio?

[Glenn Geller (VDOPh)]

Hi Mike,

Technically, it's possible. It takes a bit of work to extract the right
stream links from the playlist file, and some services hide this from
sight, best they can.

We tried to do a similar thing, streaming smoothjazz.com radio through
Askerisk, as MOH.

What we found was that the way Asterisk works (or worked originally, when
we tested), it tries to open a new socket for each MOH "session", and
they'd see multiple sessions from our IP address, and eventually block the
IP.

What we did to alleviate this, was install a locally controlled Icecast
server used as a "relay", which pulls 1 stream from them, and allows us to
pull multiple streams (relays) from the server... never been blocked since.

As far as the legality is concerned, technically ANY music for "public
consumption" requires a performance license of some kind. However, as long
as they'e not a huge company, and just using for hold music, or internal
office music... should be fine.

Good luck!



*Glenn Geller*

*VDO-Ph International*

On Wed, May 16, 2018 at 8:02 AM Mike Diehl  wrote:

> Hi all,
>
> I have a user who would like to stream their favorite radio station from
> iHeart radio for their music on hold.
>
> It this TECHNICALLY possible?  If so, any pointers would be appreciated.
>
> Is this LEGAL in the US?
>
> Thanks in advance,
>
> Mike.
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Re: [asterisk-users] Lightweight ODBC DB

[Glenn Geller (VDOPh)]

Hi Dovid,

Totally possible... We're currently doing what you suggest with standard
mySQL configured in Master-Master replication mode, across many nodes
geo-distributed, with no (or few, like environment related) errors... for a
few years now.

This allows for fast updates at any node, and synchronizes via replication
across other nodes as necessary.

There are other replication solutions out there, but nothing as free, and
"light" as this, that we've found so far.

Hope this helps.



*Glenn Geller*

*VDOTel*

On Tue, Jul 30, 2019 at 6:56 PM Dovid Bender  wrote:

> Hi,
>
> I am running several Asterisk boxes with realtime around the world. Does
> anyone have a recommendation for a "light" db that would work with Asterisk
> that would also allow replication between all sites (so if I add an entry
> to one box it will work with the rest)?
>
> TIA.
>
> Dovid
>
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