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Grant
Grant
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Hello

I am very new to ClearOS but have been a long time VoIP user with Asterisk and would greatly appreciate some guidance on a VoIP solution. I think ClearOS is the solution I need but also need to have VoIP capability, and would be even better if this was an optional module. From what I have read I think Freeswitch could be the better solution than Asterisk?. This is for residential use and want to keep the system components right down. I have followed the guide here http://www.clearfoundation.com/docs/howtos/freeswitch_with_sipxecs to install SipXecs but fails on a number of dependencies which I assume aren't available in the ClearOS repositories? The guide doesn't say what version of ClearOS it applies to? My question is should this work? I have a clean install of ClearOS 6.4 on a Dual Core Atom board. In terms of Freeswitch I think the Blue.Box front end looks like a better option? Has anyone tried this?

Thanks
Tuesday, April 30 2013, 02:13 AM
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  • Accepted Answer

    Friday, May 30 2014, 06:09 AM - #Permalink
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    Hi Luis!

    It looks like I hit the same roadblocks.

    For now, probably better to install sipXecs as a standalone server, from ISO.

    If you want it to be on the same server as ClearOS, this is where I am headed:
    https://suite.tiki.org/Jitsi#Jitsi-centric_Unified_Communication

    This is part of Tiki Suite ( https://suite.tiki.org/Tiki+Suite ) but you can just install the parts you need.

    Best regards,

    M ;-)
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  • Accepted Answer

    Luis Arano
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    Saturday, March 29 2014, 12:21 AM - #Permalink
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    hey Marc,
    I am actually trying that guide, but I have encounter a roadblock (particularly when it comes to installing sipxecs)
    Could you provide guidance?

    Thanks!
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  • Accepted Answer

    Tuesday, October 08 2013, 04:47 PM - #Permalink
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    Hi!

    Those instructions are for sipXecs 4.4.0 on ClearOS 5.x with i386.

    I just added a section for sipXecs 4.6 on ClearOS 6.x with x86_64:
    http://www.clearfoundation.com/docs/howtos/freeswitch_with_sipxecs#sipxecs_46_on_clearos_6x

    Can you give this a try?

    Thanks!

    M ;-)
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  • Accepted Answer

    Yes
    Yes
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    Sunday, July 28 2013, 07:58 PM - #Permalink
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    Running FreePBX/Asterisk on COS is certainly doable. At least on COS 5.2.

    I have been running FreePBX/Asterisk on COS for over 5 years now.
    The config has 1 PSTN trunk, 4 VOIP trunks, 2 Bluetooth GSM trunks, 2 physical VOIP phones, 5 SoftVOIP phones (IOS/Android) and 2 analog phones (connected via an ATA box). Asterisk is linked to an Icecast streamer, so the audio broadcasts are also available from 1 of the VOIP trunks.
    I have built Asterisk from source, since i wanted to use the bluetooth channel, which was experimental and only available as sourcecode at that time.

    Back then I started on Asterisk 1.4.x, my current versions are:

    - COS 5.2
    - Asterisk 1.6.0.17
    - FreePBX 2.10.1.9

    Now I'm planning for an upgrade to COS 6.x on a spare quad-core intel machine and step by step migrate the software (VOIP server, Fileserver, Backup server, Mailserver, Icecast streamer, Squeezebox server) to one or more virtual boxes on my new system.

    Although I am in doubt whether such virtual configuration will perform well enough, i will start compiling the current stable Asterisk version and see what problems pop up B) .
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  • Accepted Answer

    Grant
    Grant
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    Tuesday, May 07 2013, 04:01 AM - #Permalink
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    Brian

    Thanks for that response. It would be nice to have an easy install of Asterisk with the FreePBX web interface. I'm new to COS and get the impression there's a number of differences between 5.x and 6.x? The guides for adding VoIP to COS seem to relate only to 5.x versions? I am considering doing a 5.x install to try and get VoIP working. The hardware I'm using should be able to handle the task without an issue. Agree with the LDAP integration and was also looking to include that which got me thinking about FreeSwitch.
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  • Accepted Answer

    Brian
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    Tuesday, May 07 2013, 03:34 AM - #Permalink
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    I currently use the PBX In A Flash distribution of Asterisk and FreePBX both at home and in some of the nonprofits I volunteer in. Its a pretty straightforward implementation.

    At home I currently have it running on a dedicated single core Atom 230 and it runs great, no issues. I'll probably move it into a VM under ProxMox (have a small 2 node AMD E-350 cluster setup) shortly - more for admin purposes than anything (and to be able to redeploy the old box elsewhere).

    PIAF is basically a pretty straightforward implementation of CentOS, Asterisk, and FreePBX. Reason I mention this is that you probably could simply build Asterisk on ClearOS, and then install FreePBX and get something similar.

    In fact, there was a thread on this in this forum for COS 5.2. While I didn't actually build Asterisk and FreePBX under ClearOS, I did use information on the thread to create a small plugin for COS 5.2 that would house FreePBX within the COS web gui (nothing special), it would just simply access the PIAF system to get to FreePBX. Basically a few tweaks to what was on the thread.

    I've been thinking for a while that it would be nice to create a COS 6.x plugin to do this (ie: house FreePBX, but in a nicer way), as well as actually build Asterisk under COS. Perhaps offer this as a module in marketplace, though family, day job (which involves a lot of travel), and rusty programming skills put a hamper in this.

    But one thing I'd LOVE to see, whether Asterisk runs under COS or not, is the ability to integrate Asterisk with the LDAP directory in COS. (Zentyal actually does this, but from testing Zentyal is too buggy for me to switch to (of course I'm more familiar with COS)). Might be useful to look at some of the config that Zentyal uses though on the Asterisk side and see how one can hook this up to COS in a straight Asterisk port.

    Just a few thoughts...

    In addition, I've also thought about being able to extend the user space in COS to allow for a phone access code to be added (ie: for long distance dialing). Then when someone wants to dial, Asterisk could ask for the code, and then validate the user code and password, prior to allowing the number to be dialed. Having this in COS would make it a one stop shop for setting up user accounts. I started a thread on this 18 mo or so ago, but never had time to dive into it.
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