Which PBX do you recommend: MSS, 3CX, Axon, PiaF, Elastix

Discussions about using SIP Sorcery on your own computer/server
jvwelzen
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Post by jvwelzen » Fri Aug 07, 2009 10:08 am

I still recommend the new SipSorcery version

I think this is the best open source PBX software

Thanks Aaron and others that are making this possible

SipSorcery keeps running beter by the day

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TheFug
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Post by TheFug » Fri Aug 07, 2009 11:29 am

djon wrote:You seem to have found your solution but here's an interesting project, installing Asterisk on a US$40 router.

http://www.dslreports.com/forum/r228128 ... 0GU-router
Most of the time these are tricky projects, the Linksys WRT54GL is also such a router, it means you need to flash that router with (open source) linux software, better you make the step to the Linux OS completely...
The SipSorcery software is still in devolpment, and has only limited support for the moment, Asterisk does have the advantage, that's already some time around, and has many users/devolpers .... The MiniSipServer is handy tool, with an easy to GUI, and just the functions one might need, and has no Silverlight to work.... and i'm not a skilled programmer, and want something that doesn't need much maintenance, and does it's job......
And recently, i own a Snom820 too..... :)
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

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zaheer002
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Post by zaheer002 » Fri Aug 07, 2009 9:44 pm

djon wrote:You seem to have found your solution but here's an interesting project, installing Asterisk on a US$40 router.

http://www.dslreports.com/forum/r228128 ... 0GU-router
yes, I have heard of the Asus before. I already have a Fonera flashed with DD-WRT and it runs well for past 2 years. The Fonera is also Asterisk capable so I was tempted to run that as the server but like Fug, I am not so good at coding and the Asterisk sip profiles, extension configs scared me off a bit, so started looking at Windows based solutions. Would have been nice to have * running on the Fon though only 5 Watts max power consumption compared to 50 Watts for my P2! But I guess since it doubles up as a NAS, BitTorrent client, media streamer, it makes up for it.
I still recommend the new SipSorcery version

I think this is the best open source PBX software

Thanks Aaron and others that are making this possible

SipSorcery keeps running beter by the day
what are the minimum hardware requirements for running SS - will it run on a p2 or p3?

jvwelzen
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Joined: Thu Sep 11, 2008 1:56 pm

Post by jvwelzen » Sat Aug 08, 2009 1:45 pm

For now I am testing SS on a Pentium III 1.0 Ghz with 512 MB Ram

But maybe it's better to have a faster machine

In my experience it's using a lot of cpu

BigTex
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Joined: Sat Aug 30, 2008 9:16 pm

Post by BigTex » Tue Aug 11, 2009 3:32 pm

I'm also thinking of taking the leap to a local version of something due to the recent stability issues. I'm not complaining - Aaron is up-front that MSS/SS are experimental sights and it is an amazing service. But I'm want to expand with a stable solution for the entire family.

So I'd like to have the software running on a computer that I control. But the one I would be able to run it on is a laptop connected by Wi-Fi to my internet provider. So I won't be able to control QOS for it. For that situation, it seems to me that MSS-local would be a better solution than a flavor of Asterix. As I understand it, MSS sets up the signaling only, and the voice part is a direct connection between my ATA and my provider. So the voice part wouldn't have to route through the computer that's running MSS. Is that true? Is it also true that for an Asterix solution, the voice would go through the computer running Asterix?

Aaron
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Post by Aaron » Wed Aug 12, 2009 1:03 am

Hi BigTex,

Asterisk will always set the media path up through itself initially. However once the call is set up it is possible to get Asterisk to attempt to re-INVITE it off so that it only traverses the two end points. The only problem with that is re-INVITEs have a habit of failing if one of the end points is behind a NAT or if a certain provider has blocked them for security reasons. What ends up happening after a while is that you turn off the canreinvite option on all your Asterisk SIP accounts so to alleviate problems with audio issues.

There is another SIP server called OpenSER (now know distributed as openSIPS and Kamailio). It's more along the lines of the sipsorcery server in that its core is designed to deal only with SIP signalling and by default will not get involved in the media path. A word of warning though if you think the sipsorcery dialplans are complicated they are nothing compared to OpenSER's runtime script. Asterisk's dialplan (extensions.conf) is a lot easier to get started with but it's the other bits and pieces you need to configure for Asterisk to work properly that can get tricky, sip.conf etc.

As far as stability of the sipsorcery server goes one big step forward has been taken with a new approach to dialplan processing, see this blog post for details. It's been over a week now since that update went in and it has worked flawlessly so far. There is still one issue I'm monitoring to do with database connections timing out so there is still a chance of an outage but it's now a lot smaller than it used to be. The mysipswitch service won't be getting the update so its relaibility is unchanged from a software point of view. But it's likely to be better as a couple of users who have very "demanding" traffic are mainly using sipsorcery now.

Regards,

Aaron

BigTex
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Joined: Sat Aug 30, 2008 9:16 pm

Post by BigTex » Fri Aug 14, 2009 6:34 pm

Aaron,

Thank you very much for this great explanation! For the stability and features I'm wanting, I'm thinking that SS local version is the best option. It sounds like SS is becoming stable now so I assume a local version would be even more stable - no?

The SS server would be a 1 GHz Windows laptop with 512M memory that is connected wirelessly to my LAN and is only used for occasional email reading by a family member. I want it to also serve an ATA at a remote location (using dyndns to access my LAN). Do you think this setup would have any issues?

Thanks,

Ralph

fixup77
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Joined: Sun Jan 27, 2008 1:56 am

Post by fixup77 » Fri Aug 14, 2009 7:58 pm

3cx, Axon, asterisk... I used them all, no way back.

destinycan
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Joined: Tue Jul 14, 2009 5:48 pm

when is new version being released?

Post by destinycan » Fri Aug 14, 2009 9:47 pm

Hi Aaron,

Desperately waiting for SS version 2

fixup77
Posts: 93
Joined: Sun Jan 27, 2008 1:56 am

Post by fixup77 » Sat Aug 15, 2009 8:29 pm

Me too, I need this new Google Voice feature.

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