[asterisk-users] Summary: Upgrading to Asterisk 1.4

[Philipp von Klitzing]

Hi!

> Now over to a summary of the feedback. I'm not going deeper into bugs
> reported, those will be handled separately. 

Looks like I am a bit late, but I'll try to add my share as well to 
highlight some of the issues that are invovled with 1.2 to 1.4 
transition:

- with the advent of the "g726aal2 troubles" my preferred codec was 
rendered unusable, and it still is that way because this setup is too 
flakey, you never know if and when garbled audio will hit you. This still 
does not work cleanly between 1.2 and 1.4 Asterisk boxes, with me 
thinking that somehow on IAX this is more troublesome than on SIP. Only 
alaw/ulaw (too hungry) and gsm (too sparse) are left since ilbc has the 
potential to crash asterisk once a while (not always, not on every box).

- likewise SIP INFO DTMF worked reasonable well in Asterisk 1.2, whereas 
my experience is that in 1.4 one should better move (back) over to 
RFC2833, and when doing so don't forget about the rfc2833compensate 
setting.

- all the transitions of the type "application --> function" can be 
painful and error prone, especially for what concerns the replacements 
for DBPut and DBGet and all the levels of () and [] and {} that are now 
invovled.

- the GROUP_COUNT and call-limit (SIP) features saw a *lot* of changes on 
their path from 1.0 to 1.2 to 1.4, and I hear that for 1.6 call-limit 
will be touched and changed yet again. So practically every new point 
release does this in an entirely different fashion.

By the way, the README file in asterisk-1.4 is outdated and refer to 
upgrade instructions from 1.0 to 1.2.

Having said all of the above: Asterisk is coool and grrrrreat, and 
everyone involved even more so - Olle included ;-) - thank you for all 
the effort!

Cheers & happy days,
Philipp von Klitzing