VOIP Solution Help

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Kalvaer

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I have been trying to look though cant seem to find any details on configuring a system up. Is there something somewhere here, or is it one of those things that people dont want to talk about yet since they are making money doing it?

Basically, We have 4 incoming lines here, which go into a Telkom PABX. the first line is a hunting line, and as long as the other 3 lines are open, people only have to phone the first one to get through to our switch board?

Now I am thinking of getting rid of the Telkom PABX and setting up a VOIP gateway/PABX/. whatever

My Boss wants to keep the incoming lines for now, and route all normal calls through the lines, unless all the lines are busy, and then use VOIP, and then also use something like skype for all international calls?


Can anyone give me some advice on how to go about it? Does one simply have to take the current incoming lines out of the telkom PABX and into the VOIP PABX? what about the hunting feature, is it a function of the PABX or the Exchange, etc.

If anyone can help please let me know, as I am sure I am not the only idiot here wanting to know?
 
The hunting is a feature of the lines, not the PABX, so you can replace the PABX.

It's a small installation, so I'd suggest you look at the Yeastar MyPBX SOHO with 4x FXO ports (to connect to the Telkom lines).

Another options would be to port the Telkom number to a VoIP provider that supports Geographic Number Porting and simply go with a hosted switchboard solution.
 
Thanks Gmza, I'll look into Yeaster now.

With the existing PABX, do I just ask Telkom to come and remove it and then terminate all the incoming lines to RJ11 (or maybe at least just mark them?) I have tried to look in the current PABX but its just a spiders web of wires to me right now.

Next thing would be the "switchboard" Do you have any suggestions (will check yeastar as well) that are capable of handling the incoming calls to route them?
 
Yes, I normally recommend the SIP-T26P as the reception phone and SIP-T22P's for extensions (or SIP-T20's for extensions if you're on a tight budget).
 
OK so I got the Yeastar MyPBX and I'm currently just testing it with two phones and running it through my normal phone line into one of the FXO's

Only thing I have noticed is that there is about a 4 second delay, where it rings, before going through the phone.
I also cant seem to get the hangup right where it always seems to hold for 15 seconds before dropping a line

Anyone know or had simular situations with the yeastar and know what to set to change it, as I cant find anything?
 
Also the tone schemes country should be set to "South Africa" and caller id to v.23. There is a hangup detection tone feature
 
OK I have no clue where to change any of those other than whats available on the myPBX. I have found tons of sites giving details on how to mod things on an asterix server though not for the the yeastar things.

I see there is a V.23 for Uk and Japan that can be set, Does it matter which one? or is there a way around it.

Anyway, so far other than that above. Everything seems to be working quiet nicely. Just getting all the routes and things setup. I have another question though that I am sure somebody else has come across. I took one of the phones home, and also tried the 3CX and X-lite Soft phones. I put in the dyndns details into any of the phones and it picks up right away that the phone is working. Lsat night for about 2 min, I was able to actually send DTMF tones and so on to the mail boxes and then it all stopped. I have now tried almost everything and even have a friend trying. We can connect to the PBX, but we cant hear each other. Pretty much exactly what is explained here: http://www.voip-info.org/wiki/view/NAT+and+VOIP Though I cant seem to get any of the workarounds to work?
 
if you look at the externhost and localnets options you should be able to put in your DynDNS address there and have it set the external address correctly for NAT.

Documentation is, unfortunately, seriously lacking.
 
Hey Gmza, Where would I even begin to look for those? (FYI, Ended up getting the Yeastar Standard PBX) I cant seem to see anything like that on the webpage's config?

The phones connect to the PBX and register perfectly. When I try from outside the network though from home on my ADSL account to work. The phones ring, and even get transferred if needed, But no voice can be heard at all?
I"m trying to get it set up that I pretty much don't have to come into the office :D I can remote desktop in from home, and have all my calls transferred straight to my desk at home. So far its all working, EXCEPT for the fact that I cant hear anything?
 
We've just implemented a CentOS box with Asterisk/Freepbx. So far so good and I'm quite happy with it.

Only problem is when someone calls in, hangs up, the pabx doesn't drop the call until a minute has passed, busy browesing the other threads for help now.
 
I had a similar problem. Got it to drop now after 12 seconds if the call isn't ended right by changing enabling the busy detection, Setting the busy count to 4, and then busy pattern as 500,500.

Currently trying to get skype working now through certain dialled extensions and still cant seem to phone from an outside extension on the ADSL line ?
 
gmza said:
if you look at the externhost and localnets options
Click to expand...
Ok finally found it :o Was in systems settings under SIP settings. I put the dyndns in there as you mentioned and now I can phone outside extensions perfectly (for now at least LOL)

Next on my list if anyone can help. I have created a skype business account, and bought a number and linked an account to it. If I use skype to dial the account. The PBX gets the call and sends it through as intended, however when I phone the number (0110839983) It just rings twice and dies. Also when I try dial out using the VOIP trunk. the Yeastar tells me that all the circuits are currently busy? Anyone know what's up with that?
 
Well got 99% of it working now with SKYPE. Yeastar was really great and one of their Tech's and a Developer spent most of friday logged into my system checking things out and reconfiguring the PBX.

Right now all we have to sort out is the incoming calls from Skype. Skype is sending the incoming line as an extension and the PBX isn't able to forward it to a 10 digit extension. Will have to see what yeastar can come up with today with that as I am sure its just something with an incoming rule we need to try congfigure
 
How about a manual Asterisk + Analog card set up? TrixBox or similar will have some sort of gui's.
 
Vince0 said:
How about a manual Asterisk + Analog card set up? TrixBox or similar will have some sort of gui's.
Click to expand...
That would work, but I was hoping to use an all in one solution instead of me having to do it all manually. Which of course I have now learnt how to do anyway . It does however seem that the problem I have been having is actually a problem with the yeastar. The support guys there though have been really helpful and have discovered a bug in their firmware. They have also said they will reconfigure the PBX for me, as soon as I have phone lines again and access to my desk at work (WTF people steal fibre optic cables I still don't know???)
 
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