Looking for some PBX advice, for multiple branches

under the device options in extensions


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

I applied that setting, but it still doesn't work. Interestingly, if I use that same extension on something like Blink or X-Lite it works fine, but not on the phone itself.

also on command line

asterisk -rx 'core show codecs'
It's there:
Code:
               INT    BINARY                  HEX   TYPE       NAME   DESCRIPTION
-----------------------------------------------------------------------------------
                  1 (1 <<  0)                (0x1)  audio       g723   (G.723.1)
                  2 (1 <<  1)                (0x2)  audio        gsm   (GSM)
                  4 (1 <<  2)                (0x4)  audio       ulaw   (G.711 u-law)
                  8 (1 <<  3)                (0x8)  audio       alaw   (G.711 A-law)
                 16 (1 <<  4)               (0x10)  audio   g726aal2   (G.726 AAL2)
                 32 (1 <<  5)               (0x20)  audio      adpcm   (ADPCM)
                 64 (1 <<  6)               (0x40)  audio       slin   (16 bit Signed Linear PCM)
                128 (1 <<  7)               (0x80)  audio      lpc10   (LPC10)
                256 (1 <<  8)              (0x100)  audio       g729   (G.729A)
                512 (1 <<  9)              (0x200)  audio      speex   (SpeeX)
               1024 (1 << 10)              (0x400)  audio       ilbc   (iLBC)
               2048 (1 << 11)              (0x800)  audio       g726   (G.726 RFC3551)
               4096 (1 << 12)             (0x1000)  audio       g722   (G722)
               8192 (1 << 13)             (0x2000)  audio     siren7   (ITU G.722.1 (Siren7, licensed from Polycom))
[/quote]

and
asterisk -rx 'core show translations'

This command doesn't work:

Code:
[root@localhost ~]# asterisk -rx 'core show translations'
No such command 'core show translations' (type 'core show help core show translations' for other possible commands)

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.
Thanx, I'll check it out.
 
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As you are major guy on this site, really hope we can meet at the conference tomorrow and say hi, could allways resolve this over a coffee ! Hope you will be there ?
 
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.
Ok, it looks like the phone is working now. I can ring it from my laptop, although I'm not in the office so I can't hear it ringing, but it rang from the laptop - after installing the free g729 codex. Thanx for the headsup.


As you are major guy on this site, really hope we can meet at the conference tomorrow and say hi, could allways resolve this over a coffee ! Hope you will be there ?

I don't think I'll be able to attend tomorrow, sorry.
 
If you are going the DIY route; for what it's worth this is what currently works for me. Personally I like having full control over my PBX hardware and not be tied to any single service provider.

Using the once-off licensed g729 codecs for the sip trunk only, everything else runs g711-alaw or speex.

Having a wisp connection only for voip is beneficial because it is completely independent to Telkom, so when lines get stolen or the dsl is down calls still work. It also removes the need to have strict QoS rules and worry about data utilization impacting on calls.

For simple inter-branch operations most sip providers allow free on-net calls, so it's just a case of deploying a sip gateway with a decent internet connection. Using a sip provider as a hub rather than trunking directly to a pbx hosted yourself negates the risk of your pbx getting hacked or [D]DoS'd as it does not need to otherwise be opened to the internet.
 
If you are going the DIY route; for what it's worth this is what currently works for me. Personally I like having full control over my PBX hardware and not be tied to any single service provider.
This is one of the main reasons I decided todo it myself.


- Well, FreePBX is similar to Elastix - I honestly can't say why I chose the one over the other, but I need a web based solution for some of the other staff to be able to add / edit extensions, setup routing etc.
-
I'm not fond of Mweb, but their rates seem very good. Do you know if you can port forward a normal telkom landline to their VOIP numbers?
- Which Wifi account one do you use, and how many calls can it handle without problems?
-
I have one of those cordless phone already. It works ok'ish, and is fine for an agent won generally only takes calls but I need something that can handle 2 or 4 lines.
- I already looked at their desk phones and we still need to get some, as soon as I can convince my non-technical business partners that this is a better solution than our current Avaya PBX, which doesn't support VOIP

Using the once-off licensed g729 codecs for the sip trunk only, everything else runs g711-alaw or speex.
I installed the free g729 codec which seems to work fine as well, and from my understanding the licensing for the commercial one isn't enforced in S.A. so the free one should be fine. But it only cost $10 per license, per concurrent call so I'll purchase it if I have to.

Having a wisp connection only for voip is beneficial because it is completely independent to Telkom, so when lines get stolen or the dsl is down calls still work. It also removes the need to have strict QoS rules and worry about data utilization impacting on calls.
true, but I've had my fair share of dealings with WISP's in the past, and it wasn't great. But that was quite a few year ago, I'm sure it has changed so I'll look into this as well. Thanx for the recommendation. I actually thought of setting up a Wifi link to the closer office, which is about 5KM away, but there's almost zero visibility so I might need to setup a base station somewhere, or use a WISP, which will limit me quite a bit.

For simple inter-branch operations most sip providers allow free on-net calls, so it's just a case of deploying a sip gateway with a decent internet connection. Using a sip provider as a hub rather than trunking directly to a pbx hosted yourself negates the risk of your pbx getting hacked or [D]DoS'd as it does not need to otherwise be opened to the internet.

Agreed, but I need 2 sets of existing-in-use-telkom numbers to talk to each other, between the 2 branches, and possibly a 3rd one in Randburg.
 
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Part of Elastix is re-packaged FreePBX (also separately accessible) and all web based. Chose it because of the extra features (mainly HylaFAX). Was very easy to config and provision with.

At first I also wasn't too sure about mweb as a voip provider, but was worth the try as there is no fixed monthly / initial setup costs. Must say I am impressed by the call quality and (lack of) latency. Haven't attempted porting my Telkom numbers across as it's used for outgoing calls only so not sure if they offer that. Personally I'd go for an 0861 maxicall number from Telkom & just point it to the sip number, that way if a better provider comes along in the future it's just a case of rather painlessly updating the 0861 destination.

Regarding WISP, I'm with Amobia on the 1.5GB, 1mbit symmetric unshaped capped account for R199 p/m, haven't tried saturating it with calls, but it handles my 5 simultaneous without any problems. There is are some calculators around such as this one. Theoretically one g729 channel uses 8kbps symmetrical bandwidth, if you divide the line speed into that it should give a rough idea of what it can handle. I also chose Amobia because they break-out at Mweb. Average ping is 20ms to the sip server compared to 18ms on the business uncapped ADSL.

I use the A510IP (one base, two handsets) as it handles 2 simultaneous calls. SNOM M9 can apparently handle 4 simultaneous calls, but I find the build quality, range & battery life to be inferior by far to Siemens.

Regarding branches, there are a few options:
  • Keep existing pbx, add sip gateway to free trunk lines, use sip for least-cost routing and inter-branch routing.
  • Keep existing pbx, get a hybrid gateway (or mypbx) connect the gateway in series with the existing pbx.
  • Replace pbx with sip

First option is the cheapest and easiest to implement, if the existing pbx allows trunk-to-trunk routing it can forward calls to another destination.
Second option is more complex to setup but is still cheaper than replacing a whole existing system. As all lines will trunk through the gateway you will have full control over how to route calls.
Third option will be more expensive depending on how it's implemented, but you will have complete control over the whole system.

If you require full inter-branch routing (ability to forward a call from extension X at point A to extension Y at point B) then you will have to trunk the branches to the head-office and/or to each other. You must first look at deploying a separate WAN/VPN to keep the pbx's off the direct internet for security purposes. Depending on how big your business is something like an MPLS network with a hosted pbx (server managed and owned by you in a datacenter) may be a better option.

Alternatively if all you require to be able to do is forward a call from user at point A to reception/IVR at point B then rather keep it simple and use the free on-net calls of a sip provider. (i.e. each branch has it's own sip account with the same provider, calls forwarded between branches to their respective sip numbers are free)

G729 licensing gets you an official version of the codec, having tried both the free one and the paid for, there is definitely better performance from the paid version (it includes a benchmark utility to determine which version of the codec will work best with the architecture of the sever).

If management doesn't want to fork out to replace the existing pbx just get a gateway to interface into it (set the bri gateway's ports to NT mode) Have done this at one site and it works great, although did require some dabbling with settings on both the existing pbx and gateway to get working as required.

Have a wireless link between two offices (3km) via a high-site (also no los), using Ubiquiti nanobridge kitt and it works very well routing both voice and data.
 
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.

I hope for your sake your box is not mounted directly to the internet, with those weak passwords you will get hacked quick stix...lol
 
nah its not :P

i have diginet lines dedicated to voip on our mpls at all branches :)

The external facing boxes in the dmz on the other hand....
Fail2ban probably isnt a bad idea. Actually at the moment im testing load balancing udp port 5060 with lvs/piranha on our external facing pbx's. UDP load balancing is a real pain in the ass.
 
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Get 'Connection-telecom'. They are my current supplier and i'm quite happy with them.
Do not go the route to create your own PBX. Your are going to Hit you head to the wall and your boss will not be impressed.

The whole service is Hosted, which means that only a router and the devices are the hardware.
You can use a local loop diginet for the traffic which is relatively cheap.
I use free softphones (3cx) in android & iphones, but no blackberries.

Also, as the service is hosted, if one of your offices is down (due to power cuts or similar) the other one will stay running.

And the phone devices can run under 100mb without slowing down your network.

And the best thing is that my users can now reshuffle offices as they please. Just unplugg your phone and take it to your new office.
 
why is everyone against running their own pbx. Its essentially a glorified email server, with shinies.
 
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