Interesting point Larson
I have a high end commercial IP telephony product in Zultys. Asterisk is a bit of an enigma. It is good for the 5 to 10 user site as you say, yet it can easily handle hundreds of users and multiple sites.
I do actually clone my Asterisk configs. I learnt Asterisk with AMP, but soon realised I needed to move away from AMP to get the flexibility Asterisk offers.
With regards to support, I sell it and that is my business. Because I don’t buy anything from Digium (except G.729 licences) I understand that I can’t expect free support from them, like with Zultys, who rely on me to sell their product and accordingly support me.
Asterisk is typical of the major open source projects. It is powerful, yet incomplete. It is supported by peers and in that respect well supported for basic questions, but when it gets to touch problems, the help is harder to find. I can’t really complain, because I don’t spend much time trying to help others with their problems, although I do a bit from time to time.
I will eventually crack my specific problem, which is a basic codec related issue and not a fancy extra requirement. When I do, I will support my own sites mainly with my own knowledge base. Most ongoing support will relate to my hardware, like the Cisco gateways and I can get support through Smartnet.
On the comment from SteveC5000. I must tell you that your comments here are off topic to this thread and that your ramblings show your lack of understanding of how VOIP and likely how IP works. I don’t wish to flame you, but you have mentioned on a few posts that VOIP should not use UDP and allude that all users who work with UDP are somehow backward.
The fact is that voice is approximated when it is converted to data. It does not need to be perfectly sampled and compressed. UDP is perfect for carrying this type of data. If a packet should arrive out of order, the listener would likely never know. If a packet is lost the same applies. UDP is a very low overhead protocol that is perfect for carrying voice packets. In conjunction with clever mechanisms higher up the OSI stack, like jitter buffers, the quality of voice relayed via UDP is very good. The fact is that the voice has to make it from party A to party B in a couple of hundred milliseconds MAX! All the state information and packet reassembly in the world cannot make the data get there any quicker.
That is why UDP is used.