Looking for some PBX advice, for multiple branches

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SilverNodashi

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Hi,

We need to connect staff in 3 or 4 offices, 2 or 3 people per office
to a central phone system, where all the incoming lines are in 1
office, and the operator should be able forward calls to any extension
in any office, or the caller should be able to select a destination
from a prompt.

I'm thinking Astrisk..... but I've never worked with it, and teleco's
isn't our game really. Our current Avaya switchboard doesn't have a
VOIP module, and and it's apparently a rather expensive upgrade.

Can anyone please give me some pointers, and possible offer some
support when needed in the Alberton / JHB area?

To get started, what would I need for such a setup? We all have
Android / Blackberry handsets so I'm thinking we could use them as
VOIP handsets as well, but I'll need a few cordless (either analogue
or digital, please make a recommendation?) in 2 of the offices. All
the offices have at least a 4MB ADSL line and I'l sure we can upgrade
them all, or get extra lines if needed. the head office will have 4
telkom lines (via ISDN). These people do travel bookings so they'll be
on the phones a lot.


Any recommendations, pointers, etc would be highly appreciated.
 
You might want to look at a hosted PBX solution which should allow all your extensions to talk to each other regardless of location. Key thing with hosted will be your connectivity at the branches. ADSL is best effort so you need to decide how much down time you can allow for. I'd also not mix data and voice over the same line unless you are using something like vibe to prioritise the voice traffic. If you still wanted to host a PBX at head office your VoIP provider should be able to link you to the hosted branch extensions to make everything seamless.

As for using cellular handsets and wifi. IMO it's great for ad-hoc use, but you'll find supporting different phones with different apps on top of heavy battery drain is going to be a pain. Rather look at a good DECT solution. Siemens gigaset range is good bang for buck for small installs. Bigger installs look at Kirk/Polycom.
 
Can only endorse the advice that Morkhans has given you, hosted is the only way to go ! With regard to using DSL as your backhaul of choice, not recommended but you point out that you only have 2-3 people per branch, so DSL could handle this if you set up the QoS correctly in the router. I would gladly assist and advise you on Asterisk but in reality its overkill, host it my friend !
 
Hosted won't work in this case since the telkom number which is assigned to the ADSL lines are the active numbers as well.....
And my business partner prefers to keep the PBX "inhouse"
 
So as I've suggested you can keep your PBX at the head office and go hosted for the branches and hook them up to your PBX via the VoIP provider. You can also port your Telkom numbers to a VoIP provider if you wanted to.

If you want help with an Asterisk PBX and to look at some design options PM me your details and I'll get someone to give you a call.
 
i normally put a cisco 28xx/29xx series router and an asterisk box at each site. Pretty much any decent box is overkill. Refurbish some old servers.

into the voice router goes PRI/BRI lines from telkom. With a seperate router for the data link to the net/mpls.
This is due to my uptime requirements.
If your pbx at a branch dies, you can still route in incoming traffic from telkom to another pbx, change your option 44 dhcp tftp stuff to point elsewhere(how my phones get provisioned), and have users able to make/recieve calls.

If the telkom lines go down, i move the sharecall number to another branch and route accordingly.
If you SIP provider goes down, you still have telkom for outgoing.
If your cisco router pops, you are the unluckiest person alive. And can treat it like telkom downtime.


Might be not what you are looking for, but some ideas.
 
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I agree with Tim, the MyPBX would be your best bet, it will ease you into asterisk as it comes embedded and ready to go, if you can work a pc and have basic networking skills you can get it up and running.

I have implemented a couple of these boxes, If i remember correctly it supports openvpn so you can set up your SIP/IAX trunks securely via the tunnel and no need to open the boxes to the internet.

The guys from yeastar are also very helpful and will assist with the setup via skype, only issue is the time difference so you only have a couple of daylight hours here in SA to contact them

check out the manufacturers website http://www.yeastar.com/Products/MyPBX.asp

I buy my devices from scooop http://www.scoopdistribution.co.za/
 
Thanx guys.

Is anyone using a MyPBX, in the Alberton area, or round abouts? I'd like to see how it works if possible? I setup FreePBX but can't get the phones, or my Android, or X-Lite to communicate to the server even.
 
SoftDux-Rudi said:
Thanx guys.

Is anyone using a MyPBX, in the Alberton area, or round abouts? I'd like to see how it works if possible? I setup FreePBX but can't get the phones, or my Android, or X-Lite to communicate to the server even.
Click to expand...

You can contact our JHB branch for more information. They will also be able to give a couple of names of guys who are actively running with MyPBX.

email removed to save you from spambots
011 3129414
 
Last edited by a moderator:
Hi Guys,

So I decided to setup an Asterisk server and see if I can get this to work myself. Sofar, so-good.

I setup FreePBX 2.10.1.2 on CentOS 5.8, setup some extensions, connected my laptop and Android phone to the PBX and could make some calls between the 2 extensions.
I also got hold of a Polycom 430 SIP phone and set it up with an extension, and can phone the laptop & cellphone, but neither one can phone the Polycom - I keep getting an engaged tone.



-- Executing [s@macro-dial-one:39] Set("SIP/102-00000004", "CONNECTEDLINE(name,i)=Rudi") in new stack
-- Executing [s@macro-dial-one:40] Set("SIP/102-00000004", "CONNECTEDLINE(num)=101") in new stack
-- Executing [s@macro-dial-one:41] Set("SIP/102-00000004", "D_OPTIONS=trI") in new stack
-- Executing [s@macro-dial-one:42] Dial("SIP/102-00000004", "SIP/101,,trI") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/101
-- Connected line update to SIP/102-00000004 prevented.
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dial-one:43] ExecIf("SIP/102-00000004", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-dial-one:44] GosubIf("SIP/102-00000004", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [s@macro-dial-one:45] MacroExit("SIP/102-00000004", "") in new stack
-- Executing [s@macro-exten-vm:15] Set("SIP/102-00000004", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:16] GosubIf("SIP/102-00000004", "0?docfu,1()") in new stack
-- Executing [s@macro-exten-vm:17] GosubIf("SIP/102-00000004", "0?docfb,1()") in new stack
-- Executing [s@macro-exten-vm:18] Set("SIP/102-00000004", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:19] ExecIf("SIP/102-00000004", "0?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:20] GotoIf("SIP/102-00000004", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf("SIP/102-00000004", "0?exit,1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones("SIP/102-00000004", "congestion") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion("SIP/102-00000004", "10") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/102-00000004' in macro 'exten-vm'
== Spawn extension (from-internal, 101, 2) exited non-zero on 'SIP/102-00000004'
-- Executing [h@from-internal:1] Hangup("SIP/102-00000004", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/102-00000004'





I setup the Polycom's SIP & lines through the web interface on 192.168.0.107. I can make calls from the Polycom to one of the other extensions.



Now I also need to get the Aterisk to make & receive outside calls.
We currently have 1x ADSL & 1x ISDN 2a (bra) in this office, and another ADSL & ISDN 2a in the other office. Eventually I want to route all incoming & outgoing calls through the same PBX.

What would be my best option, as proof of concept in getting this going?
And, does anyone know where I can get a test SIP account, that I can use to test the whole setup and load say R20 for testing purposes, without doing the whole RICA / FICA process, and waiting ages to get it up and running?
 
SoftDux-Rudi said:
Hi Guys,

So I decided to setup an Asterisk server and see if I can get this to work myself. Sofar, so-good.

I setup FreePBX 2.10.1.2 on CentOS 5.8, setup some extensions, connected my laptop and Android phone to the PBX and could make some calls between the 2 extensions.
I also got hold of a Polycom 430 SIP phone and set it up with an extension, and can phone the laptop & cellphone, but neither one can phone the Polycom - I keep getting an engaged tone.
Click to expand...

odds are its your codecs. install the g729 codecs, i find its the most widely supported. alternatively specify which codecs each extension can use with teh "allow" and "deny" options in extension config.

ie. deny - all
allow - g729
 
avert said:
odds are its your codecs. install the g729 codecs, i find its the most widely supported. alternatively specify which codecs each extension can use with teh "allow" and "deny" options in extension config.

ie. deny - all
allow - g729
Click to expand...

I have g729 enabled in FreePBX > Settings > Asterisk SIP Settings > Codecs, but don't see any other place to specifically allow / deny it for a single extension.
 
Im trying to follow this and noted the following points ! As pointed out FreePBX is just a GUI not an install ! With reference to codecs as far as i understand you have yet to connect to an extenal SIP as you were requesting options, when you do this as pointed out, g729 or IAX2 ( if it can be supported) is the way to go, Asterisk requires either the loading of the Digium g729 codec or the free version ( has some issues) its not just a point of enabling it in the GUI / phones ect , it has to exist 1st ! ( or set up pass through with enabled G729 devices) But the fact you have yet to fire a SIP trunk this is not your issue. Your Polycom has G729 license installed, no free soft phone has g729 installed but will all run OK on local LAN using G711. All devices will support g711 and communicate fine on internal LAN, lets get this OK 1st, then lets look at the SIP , i can organise a free test account no problem !
 
under the device options in extensions

https://dl.dropbox.com/u/26881350/Capture.PNG

also on command line

asterisk -rx 'core show codecs'

and
asterisk -rx 'core show translations'

Make sure that g729 is there. I've always had to install it.

http://asterisk.hosting.lv/
Its literally copying the library to /usr/lib/asterisk/modules or /usr/lib64/asterisk/modules and restarting asterisk. And dont worry about this codec being free or whatnot, its good enough for my 50k phone calls a day.
 
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paulcolmer said:
Im trying to follow this and noted the following points ! As pointed out FreePBX is just a GUI not an install !
Click to expand...

Yes that's true but it keeps the settings in a MySQL DB, instead of files, so I need todo everything "the FreePBX way" for the whole system to operate properly.

paulcolmer said:
With reference to codecs as far as i understand you have yet to connect to an extenal SIP as you were requesting options, when you do this as pointed out, g729 or IAX2 ( if it can be supported) is the way to go, Asterisk requires either the loading of the Digium g729 codec or the free version ( has some issues) its not just a point of enabling it in the GUI / phones ect , it has to exist 1st ! ( or set up pass through with enabled G729 devices) But the fact you have yet to fire a SIP trunk this is not your issue. Your Polycom has G729 license installed, no free soft phone has g729 installed but will all run OK on local LAN using G711. All devices will support g711 and communicate fine on internal LAN, lets get this OK 1st, then lets look at the SIP , i can organise a free test account no problem !
Click to expand...

mmm, you raised some interesting points here:

- AFAIK g729 only needs to be licensed in the USA and is generally licensed on the server, not on the client device.
- I don't see an option to enable G711 in FreePBX, will investigate tomorrow.
- AIX is used to connect Asterisk servers, so I don't need it in this case?



Lastly, what other phones would you recommend using, instead of the Polycom? Our Avaya phones are not SIP capable so I can't use them.
 
cisco 504g, especially if you are rocking poe switches
 
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As if you read back , im so anti you do this this way ! host it ! There is no solution that can be more OK than where you are !
But you persist in going on this route and thats why you have your problems,

With regard to end points, im not in the business of marketing end points though this forum , i do not sell hardware but we have preference ! Yeahlink, AudioCodes, Snom , They work, are rubust and i dont sell them , its not hard to find out who does that ! Google ! Nothing wrong with polycom ! some of the great conference phones ever produced , not where you are now for simple endpoints !
 
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